From gabe at gundy.org  Sat Sep  1 00:30:25 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 31 Aug 2012 14:30:25 -0600
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: <CAMwZ-tL7QCCx3LiuO1rabTGB_0LHORGcF6rh3sivT=PtdTQN8w@mail.gmail.com>

On Fri, Aug 31, 2012 at 9:05 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
>>
> +1 - this is actually very handy

This is a better response to my email than I was expecting. I worried
that people would think I was grumbling about nothing. The truth is,
the mailing list archives is one of the best sources of information
about FreeSWITCH. This valuable resource gets diluted with noise
threads where people ask questions that a quick google would resolve.

The other aspect that has me thinking about participation on the list
is, the value of the audience's time. I thought I hear that there were
6,000 people on this list. If everyone takes 30 seconds to read a
single lame question, that's 50 hrs of time spent. The best way you
can show you respect your peers and the value they bring to the
community is to not waste their time.

Anyway, friends, I'm done ranting for a bit :) Again, thanks to
everyone who makes community participation so valuable!


Kind regards,
Gabe


From bdfoster at endigotech.com  Sat Sep  1 00:32:12 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 31 Aug 2012 16:32:12 -0400
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <50410E28.50904@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>
	<50410E28.50904@gmail.com>
Message-ID: <CAKV-jFyAxHgDmTUDoS17czwp0CvbfsfMbNnR8qf4KKWnUXsoDQ@mail.gmail.com>

You can indeed sort via thread. Also nabble sorts by thread much the same
but it makes it look like a forum rather than a mailing list.

+1 for next weeks topic. Its getting out of control. There are a lot of us
that contribute to this list and it really bogs us down when the same
questions keep being asked constantly. Or the questions aren't well thought
out. Most of us have full time jobs/school/families/etc. and in my case I
have two jobs and I'm going to school.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 31, 2012 3:20 PM, "Vbvbrj" <vbvbrj at gmail.com> wrote:

> 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.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep  1 00:42:00 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 13:42:00 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <5F7F4610-2849-4141-AC12-B8872C0303FC@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
Message-ID: <CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>

On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> Cool I will nail that up on my test box and see if that works
>
> Please report back on whether it works or not and then be prepared to pay
the wiki tax. :) I'll be glad to assist with getting this documented
although I think you're in the best position to give that documentation
some real-world context.

-MC
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From mike.burlingame at me.com  Sat Sep  1 00:52:21 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 31 Aug 2012 13:52:21 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
Message-ID: <071E375F-B14D-46DA-899F-38F3725F0C92@me.com>

No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)


On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:

> 
> 
> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> Cool I will nail that up on my test box and see if that works
> 
> Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
> 
> -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 freeswitch-users at vocalspace.com  Sat Sep  1 01:16:28 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Fri, 31 Aug 2012 16:16:28 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
	<CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
Message-ID: <16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>

Sorry Yes using the latest.

Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Thu Aug 30 17:17:15 2012 -0500
 
Changes made switch_cpp.cpp starting at Line 1000
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling CoreSession::destroy\n");
        if (session) {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have valid session\n");

                if (!channel) {
                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");

                        channel = switch_core_session_get_channel(session);
                }

                if (channel) {
                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
                                                          "%s destroy/unlink session from object\n", switch_channel_get_name(channel));
                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
                                                          "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
                        switch_channel_set_private(channel, "CoreSession", NULL);
                        if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel, CF_TRANSFER)) {
                                switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
                        }
                } else {
                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
                }

 [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
 [CRIT] switch_cpp.cpp:1002 We still have valid session
 [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink session from object
 [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9 destroy/unlink uuid
 [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
 [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX ) State REPORTING going to sleep
 [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX ) State Change CS_REPORTING -> CS_DESTROY
 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX  [BREAK]
 [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX ) Locked, Waiting on external entities



channel remains hung

P

On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:

> 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jpablolorenzetti at hotmail.com  Sat Sep  1 01:27:09 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 31 Aug 2012 21:27:09 +0000
Subject: [Freeswitch-users] prevent internal extensions from getting calls
	from outside
Message-ID: <DUB002-W1232DC5AC3D47DBF6757D7FD9A60@phx.gbl>

Hi, i have a gateway that allows us to call outside and allows calls to reach the extensions from outside,
but i have a bunch of extensions that we dont want to be able to receive calls from outside.
We have a block of numbers, if any of those numbers is dialed from the PSTN or any external network
the call will be routed, through the external gateway, to the PBX so it is up to the PBX to decide
if the call should reach the extension or not. there are some restrictions .. i can not have different number segments
for the extensions, they all are in the same segment, a given extension could not be allowed to be reached from outside
at this time but it may be reachable in a latter time (due to change in the user status in the company), etc ...
i tried to put a variable in the directory for a user but when the call arrives at the PBX and the PBX is routing
that variable is not available yet and i m having trouble figuring out when that variable is available before bridging the call
so i can use it as a condition to route the call.

any hint will on how to deal with will help. thanks! 		 	   		  

From anthony.minessale at gmail.com  Sat Sep  1 02:20:00 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 31 Aug 2012 17:20:00 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
	<CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
	<16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
Message-ID: <CAKbxfG8vaTvAKfTUNMV+u=jm=6isqu2AVT=DHbVWcBzkbCfmpQ@mail.gmail.com>

add this to the top of switch_core.h
#define SWITCH_DEBUG_RWLOCKS 1

rebuild and get a full log of the call
look for sign of unhandled rwlock

and put this on jira why I am i helping you over ml .... >=0


On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
<freeswitch-users at vocalspace.com> wrote:
> Sorry Yes using the latest.
>
> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Thu Aug 30 17:17:15 2012 -0500
>
> Changes made switch_cpp.cpp starting at Line 1000
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling CoreSession::destroy\n");
>         if (session) {
>         switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have valid session\n");
>
>                 if (!channel) {
>                         switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");
>
>                         channel = switch_core_session_get_channel(session);
>                 }
>
>                 if (channel) {
>                         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>                                                           "%s destroy/unlink session from object\n", switch_channel_get_name(channel));
>                         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>                                                           "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>                         switch_channel_set_private(channel, "CoreSession", NULL);
>                         if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel, CF_TRANSFER)) {
>                                 switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>                         }
>                 } else {
>                         switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
>                 }
>
>  [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>  [CRIT] switch_cpp.cpp:1002 We still have valid session
>  [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink session from object
>  [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9 destroy/unlink uuid
>  [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
>  [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX ) State REPORTING going to sleep
>  [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX ) State Change CS_REPORTING -> CS_DESTROY
>  [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX  [BREAK]
>  [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX ) Locked, Waiting on external entities
>
>
>
> channel remains hung
>
> P
>
> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>
>> 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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From freeswitch-users at vocalspace.com  Sat Sep  1 02:22:08 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Fri, 31 Aug 2012 17:22:08 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
	<CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
	<16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
Message-ID: <10DC8714-8E99-4F1C-966B-14A21C8F24E0@vocalspace.com>

Anthony,
adding switch_core_session_soft_unlock(session); below switch_core_session_rwunlock(session);


moves the call state from CS_SOFT_EXEC to CS_HANGUP but the channel still persists waiting on external entities. I am calling mod_shout to play an audio on the channel earlier in the session.

I dont know if that helps.

Thanks!
Phillip
On Aug 31, 2012, at 4:16 PM, Phillip Boles wrote:

> Sorry Yes using the latest.
> 
> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Thu Aug 30 17:17:15 2012 -0500
> 
> Changes made switch_cpp.cpp starting at Line 1000
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling CoreSession::destroy\n");
>        if (session) {
>        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have valid session\n");
> 
>                if (!channel) {
>                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");
> 
>                        channel = switch_core_session_get_channel(session);
>                }
> 
>                if (channel) {
>                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>                                                          "%s destroy/unlink session from object\n", switch_channel_get_name(channel));
>                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>                                                          "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>                        switch_channel_set_private(channel, "CoreSession", NULL);
>                        if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel, CF_TRANSFER)) {
>                                switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>                        }
>                } else {
>                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
>                }
> 
> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
> [CRIT] switch_cpp.cpp:1002 We still have valid session
> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink session from object
> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9 destroy/unlink uuid
> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX ) State REPORTING going to sleep
> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX ) State Change CS_REPORTING -> CS_DESTROY
> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX  [BREAK]
> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX ) Locked, Waiting on external entities
> 
> 
> 
> channel remains hung
> 
> P
> 
> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
> 
>> 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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mailing-lists at phoenixinternet.net  Sat Sep  1 03:51:31 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Fri, 31 Aug 2012 16:51:31 -0700
Subject: [Freeswitch-users] sofia xmlstatus profile internal reg
Message-ID: <50414E03.40800@phoenixinternet.net>

I believe I have found a bug in the response of "sofia xmlstatus profile 
internal reg". I just do not know how to report it.

It is not returning all special characters in UTF-8 format. For example 
'<' should be returned as &lt;. In the <contact> xml tag it returns a 
correct string. In the <agent> xml tag it does not return it correctly. 
Below is an example from my test server. The 3rd entry has an agent that 
returns both less than and greater than symbols and freeswitch is not 
sending them as &lt; or &gt;.

freeswitch at internal> sofia xmlstatus profile internal reg
<?xml version="1.0" encoding="ISO-8859-1"?>
<profile>
   <registrations>
     <registration>
<call-id>000ed700-06320003-0572ebcb-59776109 at 172.16.5.83</call-id>
         <user>190 at 192.168.1.6</user>
         <contact>"user" 
&lt;sip:190 at 192.168.1.90:1307;transport=udp&gt;</contact>
         <agent>Cisco-CP7940G/8.0</agent>
         <status>Registered(UDP)(unknown) exp(2012-08-31 16:42:28) 
expsecs(153)</status>
         <host>test.abc.com</host>
         <network-ip>192.168.1.90</network-ip>
         <network-port>1307</network-port>
         <sip-auth-user>190</sip-auth-user>
         <sip-auth-realm>test.abc.com</sip-auth-realm>
         <mwi-account>190 at 192.168.1.6</mwi-account>
     </registration>
     <registration>
<call-id>NTVkZWI5NDM5Yzk0ZWY4M2Q0NDcyNDE0YWM0OTJkMWE.</call-id>
         <user>106 at 192.168.1.6</user>
         <contact>"Gilbert Gutierrez" 
&lt;sip:106 at 192.168.1.90:18360;rinstance=4645682e9800585c&gt;</contact>
         <agent>X-Lite release 5.0.0 stamp 67284</agent>
         <status>Registered(UDP)(unknown) exp(2012-08-31 17:06:16) 
expsecs(1581)</status>
         <host>test.abc.com</host>
         <network-ip>192.168.1.90</network-ip>
         <network-port>18360</network-port>
         <sip-auth-user>106</sip-auth-user>
         <sip-auth-realm>test.abc.com</sip-auth-realm>
         <mwi-account>106 at 192.168.1.6</mwi-account>
     </registration>
     <registration>
<call-id>QC8jV0-xAG1nCDtN2 at test.abc.com</call-id>
         <user>180 at 192.168.1.6</user>
         <contact>PI FAX &lt;sip:180 at 192.168.1.114:5060&gt;</contact>
         <agent>Patton Smartlink 4020 <3.01.002 20 EN n0 
(1214)><00a0ba017198></agent>
         <status>Registered(UDP)(unknown) exp(2012-08-31 17:24:31) 
expsecs(2676)</status>
         <host>test.abc.com</host>
         <network-ip>192.168.1.114</network-ip>
         <network-port>5060</network-port>
         <sip-auth-user>180</sip-auth-user>
         <sip-auth-realm>test.abc.com</sip-auth-realm>
         <mwi-account>180 at 192.168.1.6</mwi-account>
     </registration>
   </registrations>
</profile>


Thank you,
Gilbert T. Gutierrez, Jr.
Operations Manager
Phoenix Internet


From msc at freeswitch.org  Sat Sep  1 07:39:37 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 20:39:37 -0700
Subject: [Freeswitch-users] prevent internal extensions from getting
 calls from outside
In-Reply-To: <DUB002-W1232DC5AC3D47DBF6757D7FD9A60@phx.gbl>
References: <DUB002-W1232DC5AC3D47DBF6757D7FD9A60@phx.gbl>
Message-ID: <CAKzWOxX8ZmsnBabAJK7iETgRcApLWyQF+=1EZySNJkr-RM5Gkw@mail.gmail.com>

How is the caller getting to the extension in the first place? Do you have
a block of DIDs or is the caller dialing an ext at an auto attn?
-MC

On Friday, August 31, 2012, Juan Pablo L. <jpablolorenzetti at hotmail.com>
wrote:
> Hi, i have a gateway that allows us to call outside and allows calls to
reach the extensions from outside,
> but i have a bunch of extensions that we dont want to be able to receive
calls from outside.
> We have a block of numbers, if any of those numbers is dialed from the
PSTN or any external network
> the call will be routed, through the external gateway, to the PBX so it
is up to the PBX to decide
> if the call should reach the extension or not. there are some
restrictions .. i can not have different number segments
> for the extensions, they all are in the same segment, a given extension
could not be allowed to be reached from outside
> at this time but it may be reachable in a latter time (due to change in
the user status in the company), etc ...
> i tried to put a variable in the directory for a user but when the call
arrives at the PBX and the PBX is routing
> that variable is not available yet and i m having trouble figuring out
when that variable is available before bridging the call
> so i can use it as a condition to route the call.
>
> any hint will on how to deal with will 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
>

-- 
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  Sat Sep  1 07:41:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 20:41:13 -0700
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <10DC8714-8E99-4F1C-966B-14A21C8F24E0@vocalspace.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
	<CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
	<16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
	<10DC8714-8E99-4F1C-966B-14A21C8F24E0@vocalspace.com>
Message-ID: <CAKzWOxXfkX4b+P0bgjRpXNjKCjJ8r-PAji2h95q8xHXseLiJwA@mail.gmail.com>

Plz get that jira opened! :)
-MC

On Friday, August 31, 2012, Phillip Boles <freeswitch-users at vocalspace.com>
wrote:
> Anthony,
> adding switch_core_session_soft_unlock(session);
below switch_core_session_rwunlock(session);
>
> moves the call state from CS_SOFT_EXEC to CS_HANGUP but the channel still
persists waiting on external entities. I am calling mod_shout to play an
audio on the channel earlier in the session.
> I dont know if that helps.
> Thanks!
> Phillip
> On Aug 31, 2012, at 4:16 PM, Phillip Boles wrote:
>
> Sorry Yes using the latest.
>
> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Thu Aug 30 17:17:15 2012 -0500
>
> Changes made switch_cpp.cpp starting at Line 1000
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling
CoreSession::destroy\n");
>        if (session) {
>        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still
have valid session\n");
>
>                if (!channel) {
>                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");
>
>                        channel = switch_core_session_get_channel(session);
>                }
>
>                if (channel) {
>
                       switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
SWITCH_LOG_DEBUG,
>                                                          "%s
destroy/unlink session from object\n", switch_channel_get_name(channel));
>
                       switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
SWITCH_LOG_DEBUG,
>                                                          "%s
destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>                        switch_channel_set_private(channel, "CoreSession",
NULL);
>                        if (switch_channel_up(channel) &&
switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel,
CF_TRANSFER)) {
>                                switch_channel_hangup(channel,
SWITCH_CAUSE_NORMAL_CLEARING);
>                        }
>                } else {
>                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
>                }
>
> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
> [CRIT] switch_cpp.cpp:1002 We still have valid session
> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink
session from object
> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9
destroy/unlink uuid
> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard
REPORTING, cause: NORMAL_CLEARING
> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX )
State REPORTING going to sleep
> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX )
State Change CS_REPORTING -> CS_DESTROY
> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX
 [BREAK]
> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX )
Locked, Waiting on external entities
>
>
>
> channel remains hung
>
> P
>
> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>
> 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)
>
> =

-- 
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  Sat Sep  1 07:47:06 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 20:47:06 -0700
Subject: [Freeswitch-users] sofia xmlstatus profile internal reg
In-Reply-To: <50414E03.40800@phoenixinternet.net>
References: <50414E03.40800@phoenixinternet.net>
Message-ID: <CAKzWOxXkr6ckS1=qPHotVeX8LhuTVLFF46dgjbsJaiUmo96o7A@mail.gmail.com>

Definitely put this on jira.freeswitch.org.
-MC

On Friday, August 31, 2012, Gilbert T. Gutierrez, Jr. <
mailing-lists at phoenixinternet.net> wrote:
> I believe I have found a bug in the response of "sofia xmlstatus profile
> internal reg". I just do not know how to report it.
>
> It is not returning all special characters in UTF-8 format. For example
> '<' should be returned as &lt;. In the <contact> xml tag it returns a
> correct string. In the <agent> xml tag it does not return it correctly.
> Below is an example from my test server. The 3rd entry has an agent that
> returns both less than and greater than symbols and freeswitch is not
> sending them as &lt; or &gt;.
>
> freeswitch at internal> sofia xmlstatus profile internal reg
> <?xml version="1.0" encoding="ISO-8859-1"?>
> <profile>
>    <registrations>
>      <registration>
> <call-id>000ed700-06320003-0572ebcb-59776109 at 172.16.5.83</call-id>
>          <user>190 at 192.168.1.6</user>
>          <contact>"user"
> &lt;sip:190 at 192.168.1.90:1307;transport=udp&gt;</contact>
>          <agent>Cisco-CP7940G/8.0</agent>
>          <status>Registered(UDP)(unknown) exp(2012-08-31 16:42:28)
> expsecs(153)</status>
>          <host>test.abc.com</host>
>          <network-ip>192.168.1.90</network-ip>
>          <network-port>1307</network-port>
>          <sip-auth-user>190</sip-auth-user>
>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>          <mwi-account>190 at 192.168.1.6</mwi-account>
>      </registration>
>      <registration>
> <call-id>NTVkZWI5NDM5Yzk0ZWY4M2Q0NDcyNDE0YWM0OTJkMWE.</call-id>
>          <user>106 at 192.168.1.6</user>
>          <contact>"Gilbert Gutierrez"
> &lt;sip:106 at 192.168.1.90:18360;rinstance=4645682e9800585c&gt;</contact>
>          <agent>X-Lite release 5.0.0 stamp 67284</agent>
>          <status>Registered(UDP)(unknown) exp(2012-08-31 17:06:16)
> expsecs(1581)</status>
>          <host>test.abc.com</host>
>          <network-ip>192.168.1.90</network-ip>
>          <network-port>18360</network-port>
>          <sip-auth-user>106</sip-auth-user>
>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>          <mwi-account>106 at 192.168.1.6</mwi-account>
>      </registration>
>      <registration>
> <call-id>QC8jV0-xAG1nCDtN2 at test.abc.com</call-id>
>          <user>180 at 192.168.1.6</user>
>          <contact>PI FAX &lt;sip:180 at 192.168.1.114:5060&gt;</contact>
>          <agent>Patton Smartlink 4020 <3.01.002 20 EN n0
> (1214)><00a0ba017198></agent>
>          <status>Registered(UDP)(unknown) exp(2012-08-31 17:24:31)
> expsecs(2676)</status>
>          <host>test.abc.com</host>
>          <network-ip>192.168.1.114</network-ip>
>          <network-port>5060</network-port>
>          <sip-auth-user>180</sip-auth-user>
>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>          <mwi-account>180 at 192.168.1.6</mwi-account>
>      </registration>
>    </registrations>
> </profile>
>
>
> Thank you,
> Gilbert T. Gutierrez, Jr.
> Operations Manager
> Phoenix Internet
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jpablolorenzetti at hotmail.com  Sat Sep  1 08:50:51 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 31 Aug 2012 22:50:51 -0600
Subject: [Freeswitch-users] prevent internal extensions from getting
 calls from outside
Message-ID: <DUB402-EAS126A96BAF0F133039421128D9A50@phx.gbl>

Hi, yes we have a block of DIDs, externally you would dial 2802002 and the gateway will send to the PBX 2002 which is the extension number. Then the PBX should decide if 2002 is allowed to get the call or it should present an IVR with an error message. Thanks.

-----Original Message-----

From: Michael Collins
Sent: 1 Sep 2012 03:41:46 GMT
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] prevent internal extensions from getting calls from outside

How is the caller getting to the extension in the first place? Do you have
a block of DIDs or is the caller dialing an ext at an auto attn?
-MC

On Friday, August 31, 2012, Juan Pablo L. <jpablolorenzetti at hotmail.com>
wrote:
> Hi, i have a gateway that allows us to call outside and allows calls to
reach the extensions from outside,
> but i have a bunch of extensions that we dont want to be able to receive
calls from outside.
> We have a block of numbers, if any of those numbers is dialed from the
PSTN or any external network
> the call will be routed, through the external gateway, to the PBX so it
is up to the PBX to decide
> if the call should reach the extension or not. there are some
restrictions .. i can not have different number segments
> for the extensions, they all are in the same segment, a given extension
could not be allowed to be reached from outside
> at this time but it may be reachable in a latter time (due to change in
the user status in the company), etc ...
> i tried to put a variable in the directory for a user but when the call
arrives at the PBX and the PBX is routing
> that variable is not available yet and i m having trouble figuring out
when that variable is available before bridging the call
> so i can use it as a condition to route the call.
>
> any hint will on how to deal with will 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
>

--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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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

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

From msc at freeswitch.org  Sat Sep  1 09:19:27 2012
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 31 Aug 2012 22:19:27 -0700
Subject: [Freeswitch-users] prevent internal extensions from getting
 calls from outside
In-Reply-To: <DUB402-EAS126A96BAF0F133039421128D9A50@phx.gbl>
References: <DUB402-EAS126A96BAF0F133039421128D9A50@phx.gbl>
Message-ID: <50419ADF.1070605@freeswitch.org>

On 8/31/2012 9:50 PM, Juan Pablo L. wrote:
> Hi, yes we have a block of DIDs, externally you would dial 2802002 and 
> the gateway will send to the PBX 2002 which is the extension number. 
> Then the PBX should decide if 2002 is allowed to get the call or it 
> should present an IVR with an error message. Thanks.
>
Try something like this:
Add the variable in the user XML configs like you did. In this example 
we'll use:
<variable name="allow_incoming_calls" value="false"/>

I'm not sure how you route the incoming DID block, but you could do 
something like this in your public.xml file or in a file in 
conf/dialplan/public/:
<extension name="Routing Inbound DIDs">
<condition field="destination_number" expression="^(280(20\d\d))$">
<action application="set_user" data="$2@${domain}"/>
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>

Then near the top of default.xml you could do something like this:
<extension name="Handle inbound exceptions">
<condition field="${allow_incoming_calls}" expression="^false$"/>
<condition field="destination_number" expression="^280(20\d\d)$">
<action application="log" data="INFO Call to extension $1 being routed 
to IVR"/>
<action application="ivr" data="my_ivr"/>
</condition>
</extension>

Naturally you'll need to build "my_ivr" or otherwise do something with 
the call. Standard disclaimer applies: I dropped this in off the top of 
my head so be sure to tinker with it if it doesn't work the first time. :)

-MC
-----Original Message-----
>
> From: Michael Collins
> Sent: 1 Sep 2012 03:41:46 GMT
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] prevent internal extensions from 
> getting calls from outside
>
> How is the caller getting to the extension in the first place? Do you 
> have a block of DIDs or is the caller dialing an ext at an auto attn?
> -MC
>
> On Friday, August 31, 2012, Juan Pablo L. 
> <jpablolorenzetti at hotmail.com <mailto:jpablolorenzetti at hotmail.com>> 
> wrote:
> > Hi, i have a gateway that allows us to call outside and allows calls 
> to reach the extensions from outside,
> > but i have a bunch of extensions that we dont want to be able to 
> receive calls from outside.
> > We have a block of numbers, if any of those numbers is dialed from 
> the PSTN or any external network
> > the call will be routed, through the external gateway, to the PBX so 
> it is up to the PBX to decide
> > if the call should reach the extension or not. there are some 
> restrictions .. i can not have different number segments
> > for the extensions, they all are in the same segment, a given 
> extension could not be allowed to be reached from outside
> > at this time but it may be reachable in a latter time (due to change 
> in the user status in the company), etc ...
> > i tried to put a variable in the directory for a user but when the 
> call arrives at the PBX and the PBX is routing
> > that variable is not available yet and i m having trouble figuring 
> out when that variable is available before bridging the call
> > so i can use it as a condition to route the call.
> >
> > any hint will on how to deal with will help. thanks!
> > 
> _________________________________________________________________________
> > 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 nathandownes at hotmail.com  Sat Sep  1 10:28:12 2012
From: nathandownes at hotmail.com (Mr Nathan Downes)
Date: Sat, 1 Sep 2012 16:28:12 +1000
Subject: [Freeswitch-users] sofia xmlstatus profile internal reg
In-Reply-To: <16a801cd87fb$ccc73200$66559600$@freeswitch.org>
References: <50414E03.40800@phoenixinternet.net>
	<16a801cd87fb$ccc73200$66559600$@freeswitch.org>
Message-ID: <BLU170-DS18E69BCC0DBB9E8222986FD7A50@phx.gbl>

Similar to http://jira.freeswitch.org/browse/FS-3971

 

Had same problem with & in the contact field

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Saturday, 1 September 2012 1:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sofia xmlstatus profile internal reg

 

Definitely put this on jira.freeswitch.org. 
-MC

On Friday, August 31, 2012, Gilbert T. Gutierrez, Jr.
<mailing-lists at phoenixinternet.net> wrote:
> I believe I have found a bug in the response of "sofia xmlstatus profile
> internal reg". I just do not know how to report it.
>
> It is not returning all special characters in UTF-8 format. For example
> '<' should be returned as &lt;. In the <contact> xml tag it returns a
> correct string. In the <agent> xml tag it does not return it correctly.
> Below is an example from my test server. The 3rd entry has an agent that
> returns both less than and greater than symbols and freeswitch is not
> sending them as &lt; or &gt;.
>
> freeswitch at internal> sofia xmlstatus profile internal reg
> <?xml version="1.0" encoding="ISO-8859-1"?>
> <profile>
>    <registrations>
>      <registration>
> <call-id>000ed700-06320003-0572ebcb-59776109 at 172.16.5.83</call-id>
>          <user>190 at 192.168.1.6</user>
>          <contact>"user"
> &lt;sip:190 at 192.168.1.90:1307;transport=udp&gt;</contact>
>          <agent>Cisco-CP7940G/8.0</agent>
>          <status>Registered(UDP)(unknown) exp(2012-08-31 16:42:28)
> expsecs(153)</status>
>          <host>test.abc.com</host>
>          <network-ip>192.168.1.90</network-ip>
>          <network-port>1307</network-port>
>          <sip-auth-user>190</sip-auth-user>
>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>          <mwi-account>190 at 192.168.1.6</mwi-account>
>      </registration>
>      <registration>
> <call-id>NTVkZWI5NDM5Yzk0ZWY4M2Q0NDcyNDE0YWM0OTJkMWE.</call-id>
>          <user>106 at 192.168.1.6</user>
>          <contact>"Gilbert Gutierrez"
> &lt;sip:106 at 192.168.1.90:18360;rinstance=4645682e9800585c&gt;</contact>
>          <agent>X-Lite release 5.0.0 stamp 67284</agent>
>          <status>Registered(UDP)(unknown) exp(2012-08-31 17:06:16)
> expsecs(1581)</status>
>          <host>test.abc.com</host>
>          <network-ip>192.168.1.90</network-ip>
>          <network-port>18360</network-port>
>          <sip-auth-user>106</sip-auth-user>
>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>          <mwi-account>106 at 192.168.1.6</mwi-account>
>      </registration>
>      <registration>
> <call-id>QC8jV0-xAG1nCDtN2 at test.abc.com</call-id>
>          <user>180 at 192.168.1.6</user>
>          <contact>PI FAX &lt;sip:180 at 192.168.1.114:5060&gt;</contact>
>          <agent>Patton Smartlink 4020 <3.01.002 20 EN n0
> (1214)><00a0ba017198></agent>
>          <status>Registered(UDP)(unknown) exp(2012-08-31 17:24:31)
> expsecs(2676)</status>
>          <host>test.abc.com</host>
>          <network-ip>192.168.1.114</network-ip>
>          <network-port>5060</network-port>
>          <sip-auth-user>180</sip-auth-user>
>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>          <mwi-account>180 at 192.168.1.6</mwi-account>
>      </registration>
>    </registrations>
> </profile>
>
>
> Thank you,
> Gilbert T. Gutierrez, Jr.
> Operations Manager
> Phoenix Internet
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Sep  1 21:36:07 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 1 Sep 2012 20:36:07 +0300
Subject: [Freeswitch-users] 0800 providers?
In-Reply-To: <CADrcvV-dQLeyfYqc5Dt_Y0JM_o6YeMLLvEMjFwiNBVNx1TtO-w@mail.gmail.com>
References: <CADrcvV-dQLeyfYqc5Dt_Y0JM_o6YeMLLvEMjFwiNBVNx1TtO-w@mail.gmail.com>
Message-ID: <CALNZuw=V6kKpx8g+d87vop+RN15NmU7JmMNM2d1zx7WGYXizxA@mail.gmail.com>

Do you mean 1-800 termination to the USA?
There are loads of free ones, here's two:

1) Our own Ken Rice -- simply send calls to $your_number at sip.tollfreegateway
.com
(e.g. sofia/external/$your_number at sip.tollfreegateway.com)
2) Alcazar -- $your_number at tollfree.alcazarnetworks.com


-Avi



On Fri, Aug 31, 2012 at 10:40 PM, Cesar Bermudez
<cesar.bermudez at gmail.com>wrote:

> Hi guys, what provider can give me termination for my users can dial 0800
> to USA.
> Best 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Sat Sep  1 21:47:21 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 01 Sep 2012 13:47:21 -0400
Subject: [Freeswitch-users] 0800 providers?
In-Reply-To: <CALNZuw=V6kKpx8g+d87vop+RN15NmU7JmMNM2d1zx7WGYXizxA@mail.gmail.com>
References: <CADrcvV-dQLeyfYqc5Dt_Y0JM_o6YeMLLvEMjFwiNBVNx1TtO-w@mail.gmail.com>
	<CALNZuw=V6kKpx8g+d87vop+RN15NmU7JmMNM2d1zx7WGYXizxA@mail.gmail.com>
Message-ID: <1605183.lnpbI2sG5q@sos>

 3) Callwithus - $your_number at tf.callwithus.com

On Saturday 01 September 2012 20:36:07 Avi Marcus wrote:
> Do you mean 1-800 termination to the USA?
> There are loads of free ones, here's two:
> 
> 1) Our own Ken Rice -- simply send calls to $your_number at sip.tollfreegateway
> .com
> (e.g. sofia/external/$your_number at sip.tollfreegateway.com)
> 2) Alcazar -- $your_number at tollfree.alcazarnetworks.com
> 
> 
> -Avi
> 
> 
> 
> On Fri, Aug 31, 2012 at 10:40 PM, Cesar Bermudez
> 
> <cesar.bermudez at gmail.com>wrote:
> > Hi guys, what provider can give me termination for my users can dial 0800
> > to USA.
> > Best 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
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.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  Sun Sep  2 02:26:04 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 01 Sep 2012 18:26:04 -0400
Subject: [Freeswitch-users] latest git does not build on my linux gentoo
Message-ID: <28769.1346538364@ccs.covici.com>

As of this commit:
commit 3f64fe91cbc720baf1341173f3eb7b9b6682d071
Author: Jeff Lenk <jeff at jefflenk.com>
Date:   Sat Sep 1 15:52:19 2012 -0500

    windows fix for libtiff on last commit

I am getting an error during the configure phase for the building of
libtiff as follows:

config.status: creating build/Makefile
config.status: error: cannot find input file: `contrib/Makefile.in'
configure: error: ./configure failed for libs/tiff-4.0.2

I did a bootstrap, copied my modules.conf and did ./configure
'--with-rundir=/var/run/freeswitch' '--enable-core-odbc' .

How can this be fixed?

Thanks.

-- 
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 rhuddleston at gmail.com  Sun Sep  2 02:54:16 2012
From: rhuddleston at gmail.com (Robert-IPhone)
Date: Sat, 1 Sep 2012 18:54:16 -0400
Subject: [Freeswitch-users] 0800 providers?
In-Reply-To: <1605183.lnpbI2sG5q@sos>
References: <CADrcvV-dQLeyfYqc5Dt_Y0JM_o6YeMLLvEMjFwiNBVNx1TtO-w@mail.gmail.com>
	<CALNZuw=V6kKpx8g+d87vop+RN15NmU7JmMNM2d1zx7WGYXizxA@mail.gmail.com>
	<1605183.lnpbI2sG5q@sos>
Message-ID: <42B75A24-963E-4EA6-8337-36CAC42C1117@gmail.com>

Happy with callwithus ;)


Sent from BETA iOS6

On Sep 1, 2012, at 1:47 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> 3) Callwithus - $your_number at tf.callwithus.com
> 
> On Saturday 01 September 2012 20:36:07 Avi Marcus wrote:
>> Do you mean 1-800 termination to the USA?
>> There are loads of free ones, here's two:
>> 
>> 1) Our own Ken Rice -- simply send calls to $your_number at sip.tollfreegateway
>> .com
>> (e.g. sofia/external/$your_number at sip.tollfreegateway.com)
>> 2) Alcazar -- $your_number at tollfree.alcazarnetworks.com
>> 
>> 
>> -Avi
>> 
>> 
>> 
>> On Fri, Aug 31, 2012 at 10:40 PM, Cesar Bermudez
>> 
>> <cesar.bermudez at gmail.com>wrote:
>>> Hi guys, what provider can give me termination for my users can dial 0800
>>> to USA.
>>> Best 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From cesar.bermudez at gmail.com  Sun Sep  2 02:57:41 2012
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Sat, 1 Sep 2012 16:57:41 -0600
Subject: [Freeswitch-users] 0800 providers?
In-Reply-To: <42B75A24-963E-4EA6-8337-36CAC42C1117@gmail.com>
References: <CADrcvV-dQLeyfYqc5Dt_Y0JM_o6YeMLLvEMjFwiNBVNx1TtO-w@mail.gmail.com>
	<CALNZuw=V6kKpx8g+d87vop+RN15NmU7JmMNM2d1zx7WGYXizxA@mail.gmail.com>
	<1605183.lnpbI2sG5q@sos>
	<42B75A24-963E-4EA6-8337-36CAC42C1117@gmail.com>
Message-ID: <CADrcvV__NY7QuQ6BqaErwSE+pqz1vp0tcdcW3WqpoZVdQmBb_g@mail.gmail.com>

thx to all guys !!!

On Sat, Sep 1, 2012 at 4:54 PM, Robert-IPhone <rhuddleston at gmail.com> wrote:

> Happy with callwithus ;)
>
>
> Sent from BETA iOS6
>
> On Sep 1, 2012, at 1:47 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> wrote:
>
> > 3) Callwithus - $your_number at tf.callwithus.com
> >
> > On Saturday 01 September 2012 20:36:07 Avi Marcus wrote:
> >> Do you mean 1-800 termination to the USA?
> >> There are loads of free ones, here's two:
> >>
> >> 1) Our own Ken Rice -- simply send calls to
> $your_number at sip.tollfreegateway
> >> .com
> >> (e.g. sofia/external/$your_number at sip.tollfreegateway.com)
> >> 2) Alcazar -- $your_number at tollfree.alcazarnetworks.com
> >>
> >>
> >> -Avi
> >>
> >>
> >>
> >> On Fri, Aug 31, 2012 at 10:40 PM, Cesar Bermudez
> >>
> >> <cesar.bermudez at gmail.com>wrote:
> >>> Hi guys, what provider can give me termination for my users can dial
> 0800
> >>> to USA.
> >>> Best 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
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Sep  2 03:38:23 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 2 Sep 2012 01:38:23 +0200
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <CAKV-jFyAxHgDmTUDoS17czwp0CvbfsfMbNnR8qf4KKWnUXsoDQ@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>
	<50410E28.50904@gmail.com>
	<CAKV-jFyAxHgDmTUDoS17czwp0CvbfsfMbNnR8qf4KKWnUXsoDQ@mail.gmail.com>
Message-ID: <1D4252F7-99B7-45BB-A6D1-54BBCB1233EF@gmail.com>

+1
Not that i'm an expert, but i do see questions over and over again...

---

David Villasmil


On Aug 31, 2012, at 22:32, Brian Foster <bdfoster at endigotech.com> wrote:

> You can indeed sort via thread. Also nabble sorts by thread much the same but it makes it look like a forum rather than a mailing list.
> 
> +1 for next weeks topic. Its getting out of control. There are a lot of us that contribute to this list and it really bogs us down when the same questions keep being asked constantly. Or the questions aren't well thought out. Most of us have full time jobs/school/families/etc. and in my case I have two jobs and I'm going to school.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Aug 31, 2012 3:20 PM, "Vbvbrj" <vbvbrj at gmail.com> wrote:
> 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.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Sep  2 06:00:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 1 Sep 2012 22:00:27 -0400
Subject: [Freeswitch-users] latest git does not build on my linux gentoo
In-Reply-To: <28769.1346538364@ccs.covici.com>
References: <28769.1346538364@ccs.covici.com>
Message-ID: <CAKV-jFzSBk7S2BiUZ=6PEmD8coGdNiQCo0m_gNbcKDDefCWV3A@mail.gmail.com>

Please report bugs to http://jira.freeswitch.org

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 1, 2012 6:27 PM, <covici at ccs.covici.com> wrote:

> As of this commit:
> commit 3f64fe91cbc720baf1341173f3eb7b9b6682d071
> Author: Jeff Lenk <jeff at jefflenk.com>
> Date:   Sat Sep 1 15:52:19 2012 -0500
>
>     windows fix for libtiff on last commit
>
> I am getting an error during the configure phase for the building of
> libtiff as follows:
>
> config.status: creating build/Makefile
> config.status: error: cannot find input file: `contrib/Makefile.in'
> configure: error: ./configure failed for libs/tiff-4.0.2
>
> I did a bootstrap, copied my modules.conf and did ./configure
> '--with-rundir=/var/run/freeswitch' '--enable-core-odbc' .
>
> How can this be fixed?
>
> Thanks.
>
> --
> 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 krice at freeswitch.org  Sun Sep  2 06:08:09 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 1 Sep 2012 21:08:09 -0500
Subject: [Freeswitch-users] latest git does not build on my linux gentoo
In-Reply-To: <CAKV-jFzSBk7S2BiUZ=6PEmD8coGdNiQCo0m_gNbcKDDefCWV3A@mail.gmail.com>
References: <28769.1346538364@ccs.covici.com>
	<CAKV-jFzSBk7S2BiUZ=6PEmD8coGdNiQCo0m_gNbcKDDefCWV3A@mail.gmail.com>
Message-ID: <5C20BFEE-CE4A-4FA4-86CD-97F2ACC1070E@freeswitch.org>


with a build log please
Ken
Sent from my iPad

On Sep 1, 2012, at 9:00 PM, Brian Foster <bdfoster at endigotech.com> wrote:

> Please report bugs to http://jira.freeswitch.org
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Sep 1, 2012 6:27 PM, <covici at ccs.covici.com> wrote:
> As of this commit:
> commit 3f64fe91cbc720baf1341173f3eb7b9b6682d071
> Author: Jeff Lenk <jeff at jefflenk.com>
> Date:   Sat Sep 1 15:52:19 2012 -0500
> 
>     windows fix for libtiff on last commit
> 
> I am getting an error during the configure phase for the building of
> libtiff as follows:
> 
> config.status: creating build/Makefile
> config.status: error: cannot find input file: `contrib/Makefile.in'
> configure: error: ./configure failed for libs/tiff-4.0.2
> 
> I did a bootstrap, copied my modules.conf and did ./configure
> '--with-rundir=/var/run/freeswitch' '--enable-core-odbc' .
> 
> How can this be fixed?
> 
> Thanks.
> 
> --
> 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 covici at ccs.covici.com  Sun Sep  2 07:28:00 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 01 Sep 2012 23:28:00 -0400
Subject: [Freeswitch-users] latest git does not build on my linux gentoo
In-Reply-To: <5C20BFEE-CE4A-4FA4-86CD-97F2ACC1070E@freeswitch.org>
References: <28769.1346538364@ccs.covici.com>
	<CAKV-jFzSBk7S2BiUZ=6PEmD8coGdNiQCo0m_gNbcKDDefCWV3A@mail.gmail.com>
	<5C20BFEE-CE4A-4FA4-86CD-97F2ACC1070E@freeswitch.org>
Message-ID: <7795.1346556480@ccs.covici.com>

OK, its done --4584.  What a pain in the neck, the only one worse is the
one Mozilla uses.  Attach should be on the first screen and there are
many other strange and quirky things about the form.

Ken Rice <krice at freeswitch.org> wrote:

> 
> with a build log please
> Ken
> Sent from my iPad
> 
> On Sep 1, 2012, at 9:00 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> 
> > Please report bugs to http://jira.freeswitch.org
> > 
> > Brian Foster
> > Endigo Computer LLC
> > 
> > Sent from a mobile device.
> > 
> > On Sep 1, 2012 6:27 PM, <covici at ccs.covici.com> wrote:
> > As of this commit:
> > commit 3f64fe91cbc720baf1341173f3eb7b9b6682d071
> > Author: Jeff Lenk <jeff at jefflenk.com>
> > Date:   Sat Sep 1 15:52:19 2012 -0500
> > 
> >     windows fix for libtiff on last commit
> > 
> > I am getting an error during the configure phase for the building of
> > libtiff as follows:
> > 
> > config.status: creating build/Makefile
> > config.status: error: cannot find input file: `contrib/Makefile.in'
> > configure: error: ./configure failed for libs/tiff-4.0.2
> > 
> > I did a bootstrap, copied my modules.conf and did ./configure
> > '--with-rundir=/var/run/freeswitch' '--enable-core-odbc' .
> > 
> > How can this be fixed?
> > 
> > Thanks.
> > 
> > --
> > 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
> 
> ----------------------------------------------------
> 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 peter.olsson at visionutveckling.se  Sun Sep  2 10:58:25 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 2 Sep 2012 06:58:25 +0000
Subject: [Freeswitch-users] latest git does not build on my linux gentoo
In-Reply-To: <7795.1346556480@ccs.covici.com>
References: <28769.1346538364@ccs.covici.com>
	<CAKV-jFzSBk7S2BiUZ=6PEmD8coGdNiQCo0m_gNbcKDDefCWV3A@mail.gmail.com>
	<5C20BFEE-CE4A-4FA4-86CD-97F2ACC1070E@freeswitch.org>,
	<7795.1346556480@ccs.covici.com>
Message-ID: <37310055-522B-4585-A72C-92E821443B78@visionutveckling.se>

There is an "attach" on the report form, so what do you exactly mean is so bad about it?

/Peter

2 sep 2012 kl. 05:40 skrev "covici at ccs.covici.com" <covici at ccs.covici.com>:

> OK, its done --4584.  What a pain in the neck, the only one worse is the
> one Mozilla uses.  Attach should be on the first screen and there are
> many other strange and quirky things about the form.
> 
> Ken Rice <krice at freeswitch.org> wrote:
> 
>> 
>> with a build log please
>> Ken
>> Sent from my iPad
>> 
>> On Sep 1, 2012, at 9:00 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> 
>>> Please report bugs to http://jira.freeswitch.org
>>> 
>>> Brian Foster
>>> Endigo Computer LLC
>>> 
>>> Sent from a mobile device.
>>> 
>>> On Sep 1, 2012 6:27 PM, <covici at ccs.covici.com> wrote:
>>> As of this commit:
>>> commit 3f64fe91cbc720baf1341173f3eb7b9b6682d071
>>> Author: Jeff Lenk <jeff at jefflenk.com>
>>> Date:   Sat Sep 1 15:52:19 2012 -0500
>>> 
>>>    windows fix for libtiff on last commit
>>> 
>>> I am getting an error during the configure phase for the building of
>>> libtiff as follows:
>>> 
>>> config.status: creating build/Makefile
>>> config.status: error: cannot find input file: `contrib/Makefile.in'
>>> configure: error: ./configure failed for libs/tiff-4.0.2
>>> 
>>> I did a bootstrap, copied my modules.conf and did ./configure
>>> '--with-rundir=/var/run/freeswitch' '--enable-core-odbc' .
>>> 
>>> How can this be fixed?
>>> 
>>> Thanks.
>>> 
>>> --
>>> 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
>> 
>> ----------------------------------------------------
>> 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
> 
> !DSPAM:5042d18a32761652469294!
> 


From kees at mroffice.org  Sun Sep  2 11:04:36 2012
From: kees at mroffice.org (Kees Varekamp)
Date: Sun, 2 Sep 2012 19:04:36 +1200
Subject: [Freeswitch-users] mod_spy on Windows
Message-ID: <CAHbVE909hJUk-1dGmA0OptWqtJ=djSNf7XmPpEKKdTP33cjmzw@mail.gmail.com>

Hi there,

I'd like to build/run mod_spy on Windows but there isn't a VS project file
for that app. Is it hard to make one? Are there known problems with this
module on Windows? Any help appreciated

kees
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From covici at ccs.covici.com  Sun Sep  2 13:14:37 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 02 Sep 2012 05:14:37 -0400
Subject: [Freeswitch-users] latest git does not build on my linux gentoo
In-Reply-To: <37310055-522B-4585-A72C-92E821443B78@visionutveckling.se>
References: <28769.1346538364@ccs.covici.com>
	<CAKV-jFzSBk7S2BiUZ=6PEmD8coGdNiQCo0m_gNbcKDDefCWV3A@mail.gmail.com>
	<5C20BFEE-CE4A-4FA4-86CD-97F2ACC1070E@freeswitch.org>,
	<7795.1346556480@ccs.covici.com>
	<37310055-522B-4585-A72C-92E821443B78@visionutveckling.se>
Message-ID: <21882.1346577277@ccs.covici.com>

I could find any attach on the reort form, maybe the form has some
accessibility problems.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> There is an "attach" on the report form, so what do you exactly mean is so bad about it?
> 
> /Peter
> 
> 2 sep 2012 kl. 05:40 skrev "covici at ccs.covici.com" <covici at ccs.covici.com>:
> 
> > OK, its done --4584.  What a pain in the neck, the only one worse is the
> > one Mozilla uses.  Attach should be on the first screen and there are
> > many other strange and quirky things about the form.
> > 
> > Ken Rice <krice at freeswitch.org> wrote:
> > 
> >> 
> >> with a build log please
> >> Ken
> >> Sent from my iPad
> >> 
> >> On Sep 1, 2012, at 9:00 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> >> 
> >>> Please report bugs to http://jira.freeswitch.org
> >>> 
> >>> Brian Foster
> >>> Endigo Computer LLC
> >>> 
> >>> Sent from a mobile device.
> >>> 
> >>> On Sep 1, 2012 6:27 PM, <covici at ccs.covici.com> wrote:
> >>> As of this commit:
> >>> commit 3f64fe91cbc720baf1341173f3eb7b9b6682d071
> >>> Author: Jeff Lenk <jeff at jefflenk.com>
> >>> Date:   Sat Sep 1 15:52:19 2012 -0500
> >>> 
> >>>    windows fix for libtiff on last commit
> >>> 
> >>> I am getting an error during the configure phase for the building of
> >>> libtiff as follows:
> >>> 
> >>> config.status: creating build/Makefile
> >>> config.status: error: cannot find input file: `contrib/Makefile.in'
> >>> configure: error: ./configure failed for libs/tiff-4.0.2
> >>> 
> >>> I did a bootstrap, copied my modules.conf and did ./configure
> >>> '--with-rundir=/var/run/freeswitch' '--enable-core-odbc' .
> >>> 
> >>> How can this be fixed?
> >>> 
> >>> Thanks.
> >>> 
> >>> --
> >>> 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
> >> 
> >> ----------------------------------------------------
> >> 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
> > 
> > !DSPAM:5042d18a32761652469294!
> > 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sun Sep  2 13:48:44 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 2 Sep 2012 09:48:44 +0000
Subject: [Freeswitch-users] latest git does not build on my linux gentoo
In-Reply-To: <21882.1346577277@ccs.covici.com>
References: <28769.1346538364@ccs.covici.com>
	<CAKV-jFzSBk7S2BiUZ=6PEmD8coGdNiQCo0m_gNbcKDDefCWV3A@mail.gmail.com>
	<5C20BFEE-CE4A-4FA4-86CD-97F2ACC1070E@freeswitch.org>,
	<7795.1346556480@ccs.covici.com>
	<37310055-522B-4585-A72C-92E821443B78@visionutveckling.se>,
	<21882.1346577277@ccs.covici.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15153B3F@cantor.std.visionutv.se>

I don't know really - never seen any problems with it, and I'm using it quite allot.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
Skickat: den 2 september 2012 11:14
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] latest git does not build on my linux gentoo

I could find any attach on the reort form, maybe the form has some
accessibility problems.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> There is an "attach" on the report form, so what do you exactly mean is so bad about it?
>
> /Peter
>
> 2 sep 2012 kl. 05:40 skrev "covici at ccs.covici.com" <covici at ccs.covici.com>:
>
> > OK, its done --4584.  What a pain in the neck, the only one worse is the
> > one Mozilla uses.  Attach should be on the first screen and there are
> > many other strange and quirky things about the form.
> >
> > Ken Rice <krice at freeswitch.org> wrote:
> >
> >>
> >> with a build log please
> >> Ken
> >> Sent from my iPad
> >>
> >> On Sep 1, 2012, at 9:00 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> >>
> >>> Please report bugs to http://jira.freeswitch.org
> >>>
> >>> Brian Foster
> >>> Endigo Computer LLC
> >>>
> >>> Sent from a mobile device.
> >>>
> >>> On Sep 1, 2012 6:27 PM, <covici at ccs.covici.com> wrote:
> >>> As of this commit:
> >>> commit 3f64fe91cbc720baf1341173f3eb7b9b6682d071
> >>> Author: Jeff Lenk <jeff at jefflenk.com>
> >>> Date:   Sat Sep 1 15:52:19 2012 -0500
> >>>
> >>>    windows fix for libtiff on last commit
> >>>
> >>> I am getting an error during the configure phase for the building of
> >>> libtiff as follows:
> >>>
> >>> config.status: creating build/Makefile
> >>> config.status: error: cannot find input file: `contrib/Makefile.in'
> >>> configure: error: ./configure failed for libs/tiff-4.0.2
> >>>
> >>> I did a bootstrap, copied my modules.conf and did ./configure
> >>> '--with-rundir=/var/run/freeswitch' '--enable-core-odbc' .
> >>>
> >>> How can this be fixed?
> >>>
> >>> Thanks.
> >>>
> >>> --
> >>> 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
> >>
> >> ----------------------------------------------------
> >> 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

--
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

!DSPAM:504321ae32766520714574!



From ntomer at newgen.co.in  Sun Sep  2 16:37:54 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Sun, 02 Sep 2012 18:07:54 +0530
Subject: [Freeswitch-users] sofia xmlstatus profile internal reg
Message-ID: <eekxxpt81rxtm4l8i3tmlkv6.1346589474352@email.android.com>


Regards

Nitin

Michael Collins <msc at freeswitch.org> wrote:

>Definitely put this on jira.freeswitch.org.
>-MC
>
>On Friday, August 31, 2012, Gilbert T. Gutierrez, Jr. <
>mailing-lists at phoenixinternet.net> wrote:
>> I believe I have found a bug in the response of "sofia xmlstatus profile
>> internal reg". I just do not know how to report it.
>>
>> It is not returning all special characters in UTF-8 format. For example
>> '<' should be returned as &lt;. In the <contact> xml tag it returns a
>> correct string. In the <agent> xml tag it does not return it correctly.
>> Below is an example from my test server. The 3rd entry has an agent that
>> returns both less than and greater than symbols and freeswitch is not
>> sending them as &lt; or &gt;.
>>
>> freeswitch at internal> sofia xmlstatus profile internal reg
>> <?xml version="1.0" encoding="ISO-8859-1"?>
>> <profile>
>>    <registrations>
>>      <registration>
>> <call-id>000ed700-06320003-0572ebcb-59776109 at 172.16.5.83</call-id>
>>          <user>190 at 192.168.1.6</user>
>>          <contact>"user"
>> &lt;sip:190 at 192.168.1.90:1307;transport=udp&gt;</contact>
>>          <agent>Cisco-CP7940G/8.0</agent>
>>          <status>Registered(UDP)(unknown) exp(2012-08-31 16:42:28)
>> expsecs(153)</status>
>>          <host>test.abc.com</host>
>>          <network-ip>192.168.1.90</network-ip>
>>          <network-port>1307</network-port>
>>          <sip-auth-user>190</sip-auth-user>
>>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>>          <mwi-account>190 at 192.168.1.6</mwi-account>
>>      </registration>
>>      <registration>
>> <call-id>NTVkZWI5NDM5Yzk0ZWY4M2Q0NDcyNDE0YWM0OTJkMWE.</call-id>
>>          <user>106 at 192.168.1.6</user>
>>          <contact>"Gilbert Gutierrez"
>> &lt;sip:106 at 192.168.1.90:18360;rinstance=4645682e9800585c&gt;</contact>
>>          <agent>X-Lite release 5.0.0 stamp 67284</agent>
>>          <status>Registered(UDP)(unknown) exp(2012-08-31 17:06:16)
>> expsecs(1581)</status>
>>          <host>test.abc.com</host>
>>          <network-ip>192.168.1.90</network-ip>
>>          <network-port>18360</network-port>
>>          <sip-auth-user>106</sip-auth-user>
>>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>>          <mwi-account>106 at 192.168.1.6</mwi-account>
>>      </registration>
>>      <registration>
>> <call-id>QC8jV0-xAG1nCDtN2 at test.abc.com</call-id>
>>          <user>180 at 192.168.1.6</user>
>>          <contact>PI FAX &lt;sip:180 at 192.168.1.114:5060&gt;</contact>
>>          <agent>Patton Smartlink 4020 <3.01.002 20 EN n0
>> (1214)><00a0ba017198></agent>
>>          <status>Registered(UDP)(unknown) exp(2012-08-31 17:24:31)
>> expsecs(2676)</status>
>>          <host>test.abc.com</host>
>>          <network-ip>192.168.1.114</network-ip>
>>          <network-port>5060</network-port>
>>          <sip-auth-user>180</sip-auth-user>
>>          <sip-auth-realm>test.abc.com</sip-auth-realm>
>>          <mwi-account>180 at 192.168.1.6</mwi-account>
>>      </registration>
>>    </registrations>
>> </profile>
>>
>>
>> Thank you,
>> Gilbert T. Gutierrez, Jr.
>> Operations Manager
>> Phoenix Internet
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Sun Sep  2 20:25:00 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 2 Sep 2012 09:25:00 -0700
Subject: [Freeswitch-users] mod_spy on Windows
In-Reply-To: <CAHbVE909hJUk-1dGmA0OptWqtJ=djSNf7XmPpEKKdTP33cjmzw@mail.gmail.com>
References: <CAHbVE909hJUk-1dGmA0OptWqtJ=djSNf7XmPpEKKdTP33cjmzw@mail.gmail.com>
Message-ID: <CAMWiMy-DPsagh=t9yPJx7qtANc3yu3BcRi2XKrsYLQLPm20iKQ@mail.gmail.com>

Submit a JIRA for this as a feature request.

On Sun, Sep 2, 2012 at 12:04 AM, Kees Varekamp <kees at mroffice.org> wrote:
> Hi there,
>
> I'd like to build/run mod_spy on Windows but there isn't a VS project file
> for that app. Is it hard to make one? Are there known problems with this
> module on Windows? Any help appreciated
>
> kees
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nbhatti at gmail.com  Sun Sep  2 20:53:14 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sun, 2 Sep 2012 19:53:14 +0300
Subject: [Freeswitch-users] cleanup of lua hangup hook after call
	disconnection
Message-ID: <CACak_yz5+qr=0nCwKuJ0q6w7oB3hzkOr_nZ7wUOP5tOVmXNWHw@mail.gmail.com>

Hi,
http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object says this
hangup hook is called at the end of channel disconnection. While the
channel is disconnected and the lua script is already exited, and since
there are no sessions at this point, (because we respond and hangup in the
dialplan sent by lua) the channel cleanup would be done by the lua garbage
collector? Or maybe I am understanding it wrong? The dialplan is served by
a lua script.

Thanks
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From cjbujold at accra.ca  Mon Sep  3 01:15:44 2012
From: cjbujold at accra.ca (Charles Bujold)
Date: Sun, 2 Sep 2012 18:15:44 -0300
Subject: [Freeswitch-users] IVR gives mod_dptools.c:1816 Unable to find menu
Message-ID: <004501cd8950$1dd3ef30$597bcd90$@accra.ca>

Strange error since I updated Freeswitch using "Make Current"   My main IVR
is not playing the long greeting but jumping directly to the exit  recording
and hang-up.  When I check the log I see that the following error:

 

2012-09-02 18:07:11.551338 [NOTICE] mod_sofia.c:2646 Pre-Answer
sofia/internal/250 at 192.168.20.153! 
2012-09-02 18:07:11.551338 [DEBUG] switch_channel.c:3057
(sofia/internal/250 at 192.168.20.153) Callstate Change RINGING -> EARLY 
2012-09-02 18:07:11.551338 [DEBUG] switch_core_session.c:778 Send signal
sofia/internal/250 at 192.168.20.153 [BREAK] 
EXECUTE sofia/internal/250 at 192.168.20.153 ivr(Welcome to Accra) 
2012-09-02 18:07:11.551338 [DEBUG] switch_core_session.c:924 Send signal
sofia/internal/250 at 192.168.20.153 [BREAK] 
2012-09-02 18:07:11.551338 [ERR] mod_dptools.c:1816 Unable to find menu 
2012-09-02 18:07:11.551338 [DEBUG] sofia.c:6051 Channel
sofia/internal/250 at 192.168.20.153 skipping state [early][183] 
EXECUTE sofia/internal/250 at 192.168.20.153
playback(/usr/local/freeswitch/recordings/Goodbye Message.wav) 
2012-09-02 18:07:11.551338 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms 
2012-09-02 18:07:11.631338 [DEBUG] switch_rtp.c:3594 Correct ip/port
confirmed. 
2012-09-02 18:07:15.771336 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/recordings/Goodbye Message.wav 
2012-09-02 18:07:15.771336 [NOTICE] switch_core_state_machine.c:249
sofia/internal/250 at 192.168.20.153 has executed the last dialplan
instruction, hanging up. 
2012-09-02 18:07:15.771336 [DEBUG] switch_channel.c:2914
(sofia/internal/250 at 192.168.20.153) Callstate Change EARLY -> HANGUP

 

 

The IVR is as follows:

 

<include>

                <!-- Accra Welcome Message -->

                <menu name="Welcome_to_Accra"

 
greet-long="/usr/local/freeswitch/recordings/Welcome Message Bilingue.wav"

 
greet-short="/usr/local/freeswitch/recordings/Welcome Message Bilingue.wav"

 
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

                                exit-sound="voicemail/vm-goodbye.wav"

                                confirm-macro=""

                                confirm-key=""

                                tts-engine="flite"

                                tts-voice="rms"

                                confirm-attempts="3"

                                timeout="10000"

                                inter-digit-timeout="2000"

                                max-failures="3"

                                max-timeouts="3"

                                digit-len="5">

                                <entry action="menu-exec-app" digits="1"
param="playback /usr/local/freeswitch/recordings/Transfer to Sales.wav"/>

                                <entry action="menu-exec-app" digits="1"
param="transfer 200 XML default"/>              <!-- Transfer to sales -->

                                <entry action="menu-exec-app" digits="2"
param="playback /usr/local/freeswitch/recordings/Transfer to Customer
Support.wav"/>

                                <entry action="menu-exec-app" digits="2"
param="transfer 7002 XML default"/>           <!-- Transfer to Customer
Service -->

                                <entry action="menu-exec-app" digits="3"
param="playback /usr/local/freeswitch/recordings/Transfer to
Administration.wav"/>

                                <entry action="menu-exec-app" digits="3"
param="voicemail default ${domain_name} 300"/> <!-- Transfer to Voicemail
-->

                                <entry action="menu-top" digits="9"
param=""/>            <!-- Start message again -->

                                <entry action="menu-exec-app"
digits="/(^\d{3,6}$)/" param="transfer $1 XML default"/>

                </menu>

</include>

 

 

Any help would be greatly appreciated.

 

Thanks

 

cjb

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From krice at freeswitch.org  Mon Sep  3 01:31:35 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 02 Sep 2012 16:31:35 -0500
Subject: [Freeswitch-users] IVR gives mod_dptools.c:1816 Unable to find
 menu
In-Reply-To: <004501cd8950$1dd3ef30$597bcd90$@accra.ca>
Message-ID: <CC693A67.42C9B%krice@freeswitch.org>

Whats the dialplan entry for this menu look like?

I?m concerned there might be an error there since your log says ?ivr(Welcome
to Accra)? note there is no _?s in there but your ivr config has the _?s in
the menu name


On 9/2/12 4:15 PM, "Charles Bujold" <cjbujold at accra.ca> wrote:

> Strange error since I updated Freeswitch using ?Make Current?   My main IVR is
> not playing the long greeting but jumping directly to the exit  recording and
> hang-up.  When I check the log I see that the following error:
>  
> 2012-09-02 18:07:11.551338 [NOTICE] mod_sofia.c:2646 Pre-Answer
> sofia/internal/250 at 192.168.20.153!
> 2012-09-02 18:07:11.551338 [DEBUG] switch_channel.c:3057
> (sofia/internal/250 at 192.168.20.153) Callstate Change RINGING -> EARLY
> 2012-09-02 18:07:11.551338 [DEBUG] switch_core_session.c:778 Send signal
> sofia/internal/250 at 192.168.20.153 [BREAK]
> EXECUTE sofia/internal/250 at 192.168.20.153 ivr(Welcome to Accra)
> 2012-09-02 18:07:11.551338 [DEBUG] switch_core_session.c:924 Send signal
> sofia/internal/250 at 192.168.20.153 [BREAK]
> 2012-09-02 18:07:11.551338 [ERR] mod_dptools.c:1816 Unable to find menu
> 2012-09-02 18:07:11.551338 [DEBUG] sofia.c:6051 Channel
> sofia/internal/250 at 192.168.20.153 skipping state [early][183]
> EXECUTE sofia/internal/250 at 192.168.20.153
> playback(/usr/local/freeswitch/recordings/Goodbye Message.wav)
> 2012-09-02 18:07:11.551338 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-09-02 18:07:11.631338 [DEBUG] switch_rtp.c:3594 Correct ip/port
> confirmed. 
> 2012-09-02 18:07:15.771336 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/recordings/Goodbye Message.wav
> 2012-09-02 18:07:15.771336 [NOTICE] switch_core_state_machine.c:249
> sofia/internal/250 at 192.168.20.153 has executed the last dialplan instruction,
> hanging up. 
> 2012-09-02 18:07:15.771336 [DEBUG] switch_channel.c:2914
> (sofia/internal/250 at 192.168.20.153) Callstate Change EARLY -> HANGUP
>  
>  
> The IVR is as follows:
>  
> <include>
>                 <!-- Accra Welcome Message -->
>                 <menu name="Welcome_to_Accra"
>                  
> greet-long="/usr/local/freeswitch/recordings/Welcome Message Bilingue.wav"
>                  
> greet-short="/usr/local/freeswitch/recordings/Welcome Message Bilingue.wav"
>                  
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>                                 exit-sound="voicemail/vm-goodbye.wav"
>                                 confirm-macro=""
>                                 confirm-key=""
>                                 tts-engine="flite"
>                                 tts-voice="rms"
>                                 confirm-attempts="3"
>                                 timeout="10000"
>                                 inter-digit-timeout="2000"
>                                 max-failures="3"
>                                 max-timeouts="3"
>                                 digit-len="5">
>                                 <entry action="menu-exec-app" digits="1"
> param="playback /usr/local/freeswitch/recordings/Transfer to Sales.wav"/>
>                                 <entry action="menu-exec-app" digits="1"
> param="transfer 200 XML default"/>              <!-- Transfer to sales -->
>                                 <entry action="menu-exec-app" digits="2"
> param="playback /usr/local/freeswitch/recordings/Transfer to Customer
> Support.wav"/>
>                                 <entry action="menu-exec-app" digits="2"
> param="transfer 7002 XML default"/>           <!-- Transfer to Customer
> Service -->
>                                 <entry action="menu-exec-app" digits="3"
> param="playback /usr/local/freeswitch/recordings/Transfer to
> Administration.wav"/>
>                                 <entry action="menu-exec-app" digits="3"
> param="voicemail default ${domain_name} 300"/> <!-- Transfer to Voicemail -->
>                                 <entry action="menu-top" digits="9" param=""/>
> <!-- Start message again -->
>                                 <entry action="menu-exec-app"
> digits="/(^\d{3,6}$)/" param="transfer $1 XML default"/>
>                 </menu>
> </include>
>  
>  
> Any help would be greatly appreciated.
>  
> Thanks
>  
> cjb
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jaybinks at gmail.com  Mon Sep  3 06:22:20 2012
From: jaybinks at gmail.com (jay binks)
Date: Mon, 3 Sep 2012 12:22:20 +1000
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <D97C254E-43A0-4A92-98D8-F8CD3F9B21A9@jerris.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>
	<D97C254E-43A0-4A92-98D8-F8CD3F9B21A9@jerris.com>
Message-ID: <CAO47Hpe_SQzSQ7d8-6zaKXXViv=+dE4Rx483-mwu=v3F+tTk1A@mail.gmail.com>

OI !!!!!

hows your beloved centos 6 working for you :)

On 31 August 2012 02:19, Michael Jerris <mike at jerris.com> wrote:

> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ntomer at newgen.co.in  Mon Sep  3 09:13:24 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Mon, 3 Sep 2012 10:43:24 +0530
Subject: [Freeswitch-users] Query about Valet Parking
In-Reply-To: <01ed01cd8790$0800e850$1802b8f0$@co.in>
References: <CALvF6vAoMBhg0N_jhiEeUFRxHdB2Saiv5iPf=uFNBmWDvxR5XQ@mail.gmail.com>
	<01ed01cd8790$0800e850$1802b8f0$@co.in>
Message-ID: <006001cd8992$d82fbc50$888f34f0$@co.in>

Hi,

 

I am building an application, in which callers will call a designated
number, will be presented with an IVR menu and then their call will be
parked at an extension. I need to call a Java API after that and pass on
Customer's IVR selection to that API. How can I do that? Any pointers?

 

Thanks

 

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 alex at jajah.com  Mon Sep  3 11:00:58 2012
From: alex at jajah.com (Alex Massover)
Date: Mon, 3 Sep 2012 10:00:58 +0300
Subject: [Freeswitch-users] Prevent A leg from hangup after bridge
	with	inbound ESL socket
In-Reply-To: <CAKzWOxXWVgGyKRSG3ZiXhm4GGf9WbqSs3ABFWcmOWqdVOAy37g@mail.gmail.com>
References: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin>
	<CAKzWOxW+jCoFfd-e0GkBDvsez4f66xqDLnzpYM6-QesJ+3Md7g@mail.gmail.com>
	<569384504C492C4580E88B5D54DFEAEA30CAFD3530@jjex01.jajah.dublin>
	<CAKzWOxXWVgGyKRSG3ZiXhm4GGf9WbqSs3ABFWcmOWqdVOAy37g@mail.gmail.com>
Message-ID: <569384504C492C4580E88B5D54DFEAEA30CB11E128@jjex01.jajah.dublin>

Hi,

We tried that,  it doesn't do a trick, the bridge app still assumes success, even if 180 is not forwarded to A leg.
But I think the problem is related to ESL socket, especially to inbound socket. I'm pretty sure that all these things work with dailplan, and as far as I remember even with outbound socket it's much easier. But looks like outbound socket bypass some of these flags, as channel is control by XML dialplan, but then is bridged by ESL API and not sure what exactly happens. I understand that hangup_after_bridge=false may not work, as it's not clear what should happen with the channel in case of inbound socket (as dialplan is ended), but  no reason for park_after_bridge=true not to park a channel.

Maybe it worth to fill a bug in JIRA about park_after_bridge=true and ESL inbound socket. I'll do some more clear tests and will fill one.

BR, Alex.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, August 30, 2012 7:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Prevent A leg from hangup after bridge with inbound ESL socket

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<mailto: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> [mailto: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

_________________________________________________________________________
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 gabe at gundy.org  Mon Sep  3 11:54:19 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 3 Sep 2012 01:54:19 -0600
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: <CAMwZ-tL0Rwz7bVEKiedYpWwJEs-hgTS=XygiHp5hKuNK3LVi4w@mail.gmail.com>

On Fri, Aug 17, 2012 at 10:17 AM, Jakub Tencl <info at pripojtese.net> wrote:
> 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?

Are you talking about inbound calls from your provider or outbound
calls to your provider? Just to be clear.

Gabe


From gabe at gundy.org  Mon Sep  3 11:59:34 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 3 Sep 2012 01:59:34 -0600
Subject: [Freeswitch-users] Limitation
In-Reply-To: <502B8B67.3090306@pripojtese.net>
References: <502B8B67.3090306@pripojtese.net>
Message-ID: <CAMwZ-tJdeDXKt3qEBZaqSSgkfZBqKS0+bDwugewqLGrkyc_5mg@mail.gmail.com>

On Wed, Aug 15, 2012 at 5:43 AM, Jakub Tencl <info at pripojtese.net> wrote:
> 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?

Wouldn't the limitation be on the vBilling side (if there is any)? It
doesn't seem like this list is the right place to ask. Anyway, good
luck and let us know what you learn.


Gabe


From chris at ghosttelecom.com  Mon Sep  3 12:13:30 2012
From: chris at ghosttelecom.com (Chris Martineau)
Date: Mon, 3 Sep 2012 08:13:30 +0000
Subject: [Freeswitch-users] Codec list
Message-ID: <F839D66EC0BCB04CB246EA21B8B4AF6B5B479B@DB3PRD0710MB369.eurprd07.prod.outlook.com>

Hi,

Reading the wiki/codec page I am slightly confused by what the capabilities actually are.

If I wish to use a speex 8000 codec with a 40ms ptime at the client would I just set speex at 8000@40i in the codec preferences in order to handle this? Do I just apply the @xi format to a codec in order to change its ptime to whatever value is needed or are only certain settings available as hinted at in the transcodable codecs section?

Many thanks


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From b2m at a-cti.com  Mon Sep  3 13:15:24 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Mon, 3 Sep 2012 14:45:24 +0530
Subject: [Freeswitch-users] IP trunk without registration to the carrier
In-Reply-To: <CAMwZ-tL0Rwz7bVEKiedYpWwJEs-hgTS=XygiHp5hKuNK3LVi4w@mail.gmail.com>
References: <001401cd7c62$3c1f35d0$b45da170$@com>
	<CAKzWOxXhnxSjPphJ9Zm-KeJFFWBEtt6QK2wkix0xRTLbB9OwVA@mail.gmail.com>
	<502E6EA3.4090104@pripojtese.net>
	<CAMwZ-tL0Rwz7bVEKiedYpWwJEs-hgTS=XygiHp5hKuNK3LVi4w@mail.gmail.com>
Message-ID: <CAP2fbbeLhMgXqRJXtym7gn_Z7mUxsJea0Sri=nAb=XmT681MMg@mail.gmail.com>

Just make it false (@ your service provider XML). I did the same it worked.

<param name="register" value="false"/>

Thanks,
Bala

On Mon, Sep 3, 2012 at 1:24 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Aug 17, 2012 at 10:17 AM, Jakub Tencl <info at pripojtese.net> wrote:
> > 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?
>
> Are you talking about inbound calls from your provider or outbound
> calls to your provider? Just to be clear.
>
> 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 peter.olsson at visionutveckling.se  Mon Sep  3 13:28:22 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 Sep 2012 09:28:22 +0000
Subject: [Freeswitch-users] Prevent A leg from hangup after
	bridge	with	inbound ESL socket
In-Reply-To: <569384504C492C4580E88B5D54DFEAEA30CB11E128@jjex01.jajah.dublin>
References: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin>
	<CAKzWOxW+jCoFfd-e0GkBDvsez4f66xqDLnzpYM6-QesJ+3Md7g@mail.gmail.com>
	<569384504C492C4580E88B5D54DFEAEA30CAFD3530@jjex01.jajah.dublin>
	<CAKzWOxXWVgGyKRSG3ZiXhm4GGf9WbqSs3ABFWcmOWqdVOAy37g@mail.gmail.com>,
	<569384504C492C4580E88B5D54DFEAEA30CB11E128@jjex01.jajah.dublin>
Message-ID: <1FFF97C269757C458224B7C895F35F15153DF1@cantor.std.visionutv.se>

try park_after_bridge=true, it should park the call again after the bridge, and keep it alive.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Alex Massover [alex at jajah.com]
Skickat: den 3 september 2012 09:00
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Prevent A leg from hangup after bridge with inbound ESL socket

Hi,

We tried that,  it doesn't do a trick, the bridge app still assumes success, even if 180 is not forwarded to A leg.
But I think the problem is related to ESL socket, especially to inbound socket. I'm pretty sure that all these things work with dailplan, and as far as I remember even with outbound socket it's much easier. But looks like outbound socket bypass some of these flags, as channel is control by XML dialplan, but then is bridged by ESL API and not sure what exactly happens. I understand that hangup_after_bridge=false may not work, as it's not clear what should happen with the channel in case of inbound socket (as dialplan is ended), but  no reason for park_after_bridge=true not to park a channel.

Maybe it worth to fill a bug in JIRA about park_after_bridge=true and ESL inbound socket. I'll do some more clear tests and will fill one.

BR, Alex.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, August 30, 2012 7:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Prevent A leg from hangup after bridge with inbound ESL socket

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<mailto: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> [mailto: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

_________________________________________________________________________
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

!DSPAM:504453d832762136219851!


From gvvsubhashkumar at gmail.com  Mon Sep  3 17:02:05 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 3 Sep 2012 06:02:05 -0700
Subject: [Freeswitch-users] FreeSwitch SIP Registration
Message-ID: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>

Hi All,

          I am trying to register  to freeswitch using X-lite i have done
necessary configuration on X-lite when the X-lite tries to REGISTER to
freeswitch it is saying 403 forbidden my sofia conf xml is like below  i
followed the wiki page to create the sofia configuration file.I dont know
the procedure that i followed to register the user is correct or not.Please
suggest me how can i do the sip user registeration
<profile name="My Name">
 <gateways>
 </gateways>
 <aliases>
 </aliases>
 <domains>
  <domain name="local_ip" alias="false" parse="true"/>
  <settings>
  <param name="challenge-realm" value="auto_from"/>
  <param name="force-register-domain" value="local_ip"/>
  <param name="force-subscription-domain" value="local_ip"/>
  <param name="send-message-query-on-register" value="false"/>
 </settings>
</profile>



Thanks,
Subhash.
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From stkn at openisdn.net  Mon Sep  3 17:26:14 2012
From: stkn at openisdn.net (Stefan Knoblich)
Date: Mon, 03 Sep 2012 15:26:14 +0200
Subject: [Freeswitch-users] FreeSwitch SIP Registration
In-Reply-To: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
References: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
Message-ID: <5044AFF6.2020501@openisdn.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 03.09.2012 15:02, Subhash wrote:
> Hi All,
> 
> I am trying to register  to freeswitch using X-lite i have done necessary configuration on X-lite when the X-lite tries to REGISTER to freeswitch it
> is saying 403 forbidden my sofia conf xml is like below  i followed the wiki page to create the sofia configuration file.I dont know the procedure
> that i followed to register the user is correct or not.Please suggest me how can i do the sip user registeration
> 
> <profile name="My Name"> <gateways> </gateways> <aliases> </aliases> <domains> <domain name="local_ip" alias="false" parse="true"/> <settings> <param
> name="challenge-realm" value="auto_from"/> <param name="force-register-domain" value="local_ip"/> <param name="force-subscription-domain"
> value="local_ip"/> <param name="send-message-query-on-register" value="false"/> </settings> </profile>

http://www.comicsanscriminal.com/

I can deal with the occasional 3+ colors, multiple-font HTML email, but...

Comic Sans, seriously?
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.19 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://www.enigmail.net/

iEYEARECAAYFAlBEr/YACgkQjiIIAK4rYUrTsACeLr0AYVMdaQnATYYFan2RLi44
vrAAnRH3YcrAi0EAYvYYJa/ncYHRL2y/
=jk8t
-----END PGP SIGNATURE-----


From jpablolorenzetti at hotmail.com  Mon Sep  3 18:30:08 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 3 Sep 2012 14:30:08 +0000
Subject: [Freeswitch-users] prevent internal extensions from getting
 calls from outside
In-Reply-To: <50419ADF.1070605@freeswitch.org>
References: <DUB402-EAS126A96BAF0F133039421128D9A50@phx.gbl>,
	<50419ADF.1070605@freeswitch.org>
Message-ID: <DUB002-W768A46A9BFD753B56E797ED9AB0@phx.gbl>

i followed your instructions and this is exactly what we needed,
thank you very much.


________________________________
> Date: Fri, 31 Aug 2012 22:19:27 -0700 
> From: msc at freeswitch.org 
> To: freeswitch-users at lists.freeswitch.org 
> Subject: Re: [Freeswitch-users] prevent internal extensions from  
> getting calls from outside 
>  
> On 8/31/2012 9:50 PM, Juan Pablo L. wrote: 
> Hi, yes we have a block of DIDs, externally you would dial 2802002 and  
> the gateway will send to the PBX 2002 which is the extension number.  
> Then the PBX should decide if 2002 is allowed to get the call or it  
> should present an IVR with an error message. Thanks. 
>  
> Try something like this: 
> Add the variable in the user XML configs like you did. In this example  
> we'll use: 
> <variable name="allow_incoming_calls" value="false"/> 
>  
> I'm not sure how you route the incoming DID block, but you could do  
> something like this in your public.xml file or in a file in  
> conf/dialplan/public/: 
> <extension name="Routing Inbound DIDs"> 
>    <condition field="destination_number" expression="^(280(20\d\d))$"> 
>      <action application="set_user" data="$2@${domain}"/> 
>      <action application="transfer" data="$1 XML default"/> 
>    </condition> 
> </extension> 
>  
> Then near the top of default.xml you could do something like this: 
> <extension name="Handle inbound exceptions"> 
>    <condition field="${allow_incoming_calls}" expression="^false$"/> 
>    <condition field="destination_number" expression="^280(20\d\d)$"> 
>      <action application="log" data="INFO Call to extension $1 being  
> routed to IVR"/> 
>      <action application="ivr" data="my_ivr"/> 
>    </condition> 
> </extension> 
>  
> Naturally you'll need to build "my_ivr" or otherwise do something with  
> the call. Standard disclaimer applies: I dropped this in off the top of  
> my head so be sure to tinker with it if it doesn't work the first time.  
> :) 
>  
> -MC 
> -----Original Message----- 
>  
> From: Michael Collins 
> Sent: 1 Sep 2012 03:41:46 GMT 
> To: FreeSWITCH Users Help 
> Subject: Re: [Freeswitch-users] prevent internal extensions from  
> getting calls from outside 
>  
> How is the caller getting to the extension in the first place? Do you  
> have a block of DIDs or is the caller dialing an ext at an auto attn? 
> -MC 
>  
> On Friday, August 31, 2012, Juan Pablo L.  
> <jpablolorenzetti at hotmail.com<mailto:jpablolorenzetti at hotmail.com>>  
> wrote: 
>  > Hi, i have a gateway that allows us to call outside and allows calls  
> to reach the extensions from outside, 
>  > but i have a bunch of extensions that we dont want to be able to  
> receive calls from outside. 
>  > We have a block of numbers, if any of those numbers is dialed from  
> the PSTN or any external network 
>  > the call will be routed, through the external gateway, to the PBX so  
> it is up to the PBX to decide 
>  > if the call should reach the extension or not. there are some  
> restrictions .. i can not have different number segments 
>  > for the extensions, they all are in the same segment, a given  
> extension could not be allowed to be reached from outside 
>  > at this time but it may be reachable in a latter time (due to change  
> in the user status in the company), etc ... 
>  > i tried to put a variable in the directory for a user but when the  
> call arrives at the PBX and the PBX is routing 
>  > that variable is not available yet and i m having trouble figuring  
> out when that variable is available before bridging the call 
>  > so i can use it as a condition to route the call. 
>  > 
>  > any hint will on how to deal with will help. thanks! 
>  > _________________________________________________________________________ 
>  > 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  
> 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 		 	   		  

From don.dawson at voice-ring.com  Mon Sep  3 19:29:17 2012
From: don.dawson at voice-ring.com (Don Dawson)
Date: Mon, 03 Sep 2012 10:29:17 -0500
Subject: [Freeswitch-users] SLA call on hold
Message-ID: <5044CCCD.2080502@voice-ring.com>


With the 1.2.0 freeswitch version we have the following problem:

Phone 10 calls SLA ext 122, answer the line from one of the 3 phones 
that have SLA(122) then put call on hold, all is good with sla. If one 
picks up the line(122) from another phone it hangs up.
This does work on the following version:
        FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)

Anybody tried this on the 1.2.0 version?



From asaad2 at gmail.com  Mon Sep  3 21:03:50 2012
From: asaad2 at gmail.com (BookBag)
Date: Mon, 3 Sep 2012 13:03:50 -0400
Subject: [Freeswitch-users] FreeSwitch SIP Registration
In-Reply-To: <5044AFF6.2020501@openisdn.net>
References: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
	<5044AFF6.2020501@openisdn.net>
Message-ID: <CAAdTMa8x0v5sYDrHZpqO44-uf-EV9xpv6HE_Nt1sP6FTwakTQQ@mail.gmail.com>

Check your acl and always reload sofia after any change. Also make sure you
have your firewall off for testing
On Sep 3, 2012 9:30 AM, "Stefan Knoblich" <stkn at openisdn.net> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 03.09.2012 15:02, Subhash wrote:
> > Hi All,
> >
> > I am trying to register  to freeswitch using X-lite i have done
> necessary configuration on X-lite when the X-lite tries to REGISTER to
> freeswitch it
> > is saying 403 forbidden my sofia conf xml is like below  i followed the
> wiki page to create the sofia configuration file.I dont know the procedure
> > that i followed to register the user is correct or not.Please suggest me
> how can i do the sip user registeration
> >
> > <profile name="My Name"> <gateways> </gateways> <aliases> </aliases>
> <domains> <domain name="local_ip" alias="false" parse="true"/> <settings>
> <param
> > name="challenge-realm" value="auto_from"/> <param
> name="force-register-domain" value="local_ip"/> <param
> name="force-subscription-domain"
> > value="local_ip"/> <param name="send-message-query-on-register"
> value="false"/> </settings> </profile>
>
> http://www.comicsanscriminal.com/
>
> I can deal with the occasional 3+ colors, multiple-font HTML email, but...
>
> Comic Sans, seriously?
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v2.0.19 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://www.enigmail.net/
>
> iEYEARECAAYFAlBEr/YACgkQjiIIAK4rYUrTsACeLr0AYVMdaQnATYYFan2RLi44
> vrAAnRH3YcrAi0EAYvYYJa/ncYHRL2y/
> =jk8t
> -----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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vetali100 at gmail.com  Mon Sep  3 21:28:03 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Mon, 3 Sep 2012 10:28:03 -0700
Subject: [Freeswitch-users] FreeSwitch SIP Registration
In-Reply-To: <CAAdTMa8x0v5sYDrHZpqO44-uf-EV9xpv6HE_Nt1sP6FTwakTQQ@mail.gmail.com>
References: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
	<5044AFF6.2020501@openisdn.net>
	<CAAdTMa8x0v5sYDrHZpqO44-uf-EV9xpv6HE_Nt1sP6FTwakTQQ@mail.gmail.com>
Message-ID: <CACc3D0XFPY=TDxpkYXUvBJ2ZbArxc+gk2AMzuSCT_6pZZApZ-Q@mail.gmail.com>

You need also to create users and set their passwords in the freeswitch's
/conf/directory folder.
One of the easiest way to learn this part is to use the default
configuration and modify per your needs - it already has ~20 users in
./conf/directory - so it worth to use as example.

Vitalie



2012/9/3 BookBag <asaad2 at gmail.com>

> Check your acl and always reload sofia after any change. Also make sure
> you have your firewall off for testing
> On Sep 3, 2012 9:30 AM, "Stefan Knoblich" <stkn at openisdn.net> wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> On 03.09.2012 15:02, Subhash wrote:
>> > Hi All,
>> >
>> > I am trying to register  to freeswitch using X-lite i have done
>> necessary configuration on X-lite when the X-lite tries to REGISTER to
>> freeswitch it
>> > is saying 403 forbidden my sofia conf xml is like below  i followed the
>> wiki page to create the sofia configuration file.I dont know the procedure
>> > that i followed to register the user is correct or not.Please suggest
>> me how can i do the sip user registeration
>> >
>> > <profile name="My Name"> <gateways> </gateways> <aliases> </aliases>
>> <domains> <domain name="local_ip" alias="false" parse="true"/> <settings>
>> <param
>> > name="challenge-realm" value="auto_from"/> <param
>> name="force-register-domain" value="local_ip"/> <param
>> name="force-subscription-domain"
>> > value="local_ip"/> <param name="send-message-query-on-register"
>> value="false"/> </settings> </profile>
>>
>> http://www.comicsanscriminal.com/
>>
>> I can deal with the occasional 3+ colors, multiple-font HTML email, but...
>>
>> Comic Sans, seriously?
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v2.0.19 (GNU/Linux)
>> Comment: Using GnuPG with Mozilla - http://www.enigmail.net/
>>
>> iEYEARECAAYFAlBEr/YACgkQjiIIAK4rYUrTsACeLr0AYVMdaQnATYYFan2RLi44
>> vrAAnRH3YcrAi0EAYvYYJa/ncYHRL2y/
>> =jk8t
>> -----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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Sep  3 21:46:34 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 3 Sep 2012 11:46:34 -0600
Subject: [Freeswitch-users] FreeSwitch SIP Registration
In-Reply-To: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
References: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
Message-ID: <CAMwZ-tL35_N2tuMGBoyqSEuZ_T0MgybBaoVKaWoq5SiQ8pR56A@mail.gmail.com>

On Mon, Sep 3, 2012 at 7:02 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:
> I am trying to register  to freeswitch using X-lite i have done necessary
> configuration on X-lite when the X-lite tries to REGISTER to freeswitch it
> is saying 403 forbidden my sofia conf xml is like below  i followed the wiki
> page to create the sofia configuration file.I dont know the procedure that i
> followed to register the user is correct or not.Please suggest me how can i
> do the sip user registration

Have you tried starting with the default configs and making changes from there?



Best,
Gabe


From shakad at gtt.co.gy  Mon Sep  3 21:59:42 2012
From: shakad at gtt.co.gy (Shaka Dow)
Date: Mon, 3 Sep 2012 13:59:42 -0400
Subject: [Freeswitch-users] set_zombie_exec
Message-ID: <001c01cd89fd$e5374bf0$afa5e3d0$@gtt.co.gy>

Dear FreeSWITCH users,

I have a very simple python script that is called by an extension when
dialed. But despite all my attempts on resolving the issue, even after
trying various versions (1.2.1 and the latest GIT clone) I still keep
receiving the following messages at the console when I hangup the call while
playback is being executed:

 

2012-09-03 12:00:24.706709 [DEBUG] switch_core_session.c:2357
sofia/internal/1000 at 192.168.0.121 Channel is hungup and application
'playback' does not have the zombie_exec flag

 

I tried setting the zombie_exec flag but that did not help either. This
problem seems to also occur when I use Lua.

 

Here is my extension:

 

   <extension name="test1_1">

      <condition field="destination_number" expression="^1000$">

                <action application="set_zombie_exec"/>

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

      </condition>

    </extension>

 

Here is the python script:

 

import freeswitch

 

def handler(session, args):

    session.answer()

    session.execute("set_zombie_exec")

    session.sleep(1000)

    session.execute("playback","test1/welcome.wav")

    session.hangup()

 

 

PLEASE HELP!!!!

 

Kind regards.

 

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From kris at kriskinc.com  Mon Sep  3 22:41:48 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 3 Sep 2012 14:41:48 -0400
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: <CAKDfjgcMA7e3Z22=3wcLWVyUz-Ygb3oejmRwGqfc=w_n1wiKJQ@mail.gmail.com>

Anymore info on specific traffic patterns that cause this?  I've been
beating up FS on 6.3 for several days now and haven't seen anything
unusual yet...

On Thu, Aug 30, 2012 at 11:45 PM, Ken Rice <krice at freeswitch.org> wrote:
> 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...
>
>

-- 
Kristian Kielhofner


From krice at freeswitch.org  Mon Sep  3 23:01:37 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 03 Sep 2012 14:01:37 -0500
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAKDfjgcMA7e3Z22=3wcLWVyUz-Ygb3oejmRwGqfc=w_n1wiKJQ@mail.gmail.com>
Message-ID: <CC6A68C1.43C8F%krice@freeswitch.org>

Hey Kris,

Can you grab the oprofile reports from that box and send them to me offlist?

K


On 9/3/12 1:41 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:

> Anymore info on specific traffic patterns that cause this?  I've been
> beating up FS on 6.3 for several days now and haven't seen anything
> unusual yet...
> 
> On Thu, Aug 30, 2012 at 11:45 PM, Ken Rice <krice at freeswitch.org> wrote:
>> 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...
>> 
>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From grcamauer at gmail.com  Mon Sep  3 23:25:58 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Mon, 3 Sep 2012 16:25:58 -0300
Subject: [Freeswitch-users] Missing libtiff after make current
Message-ID: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>

Today, after issuing "make current", span_dsp wouldn't compile.  Since I
had problems with this before, I issued "make spandsp-reconf" which fixed
the problem last time, but not this time.
While researching into the matter, I tried "git checkout --
libs/spandsp/INSTALL" - followed by "make current" with no joy.  I also
tried "git pull && ./bootstrap.sh && ./configure && make" to no avail.
All of these where suggestions found in other threads of this list.
I also tried "make sure".  After these attempts, I am getting the following
error upon freeswitch startup:

2012-09-04 15:52:51.286376 [CRIT] switch_loadable_module.c:1310 Error
Loading module /usr/local/freeswitch/mod/mod_spandsp.so
**libtiff.so.5: cannot open shared object file: No such file or directory**.

This is on:
CentOS 6.3 Final
Linux Freeswitch 2.6.32-279.5.2.el6.x86_64 #1 SMP Fri Aug 24 01:07:11 UTC
2012 x86_64 x86_64 x86_64 GNU/Linux
FreeSWITCH Version 1.3.0+git~20120903T183548Z~f012b7abf8 (1.3.0; git at
commit f012b7abf8 on Mon, 03 Sep 2012 18:35:48 Z) Started.
Any tips will be appreciated.
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From nickolayr at gmail.com  Tue Sep  4 00:55:07 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Mon, 3 Sep 2012 16:55:07 -0400
Subject: [Freeswitch-users] Reload dingalings jingle_profiles
Message-ID: <CAAQCs32yHMPcBOYoBwtq-db7pCFoqSC94tJPRubq_gC8GzQ4ow@mail.gmail.com>

Hello,
Does anybody know is it possible to reload *dingalings
jingle_profiles*from console?

PS: *reloadxml *and *dingalings reload* did not helps

Thanks.
--
Rogoshchenkov Nikolay
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From curriegrad2004 at gmail.com  Tue Sep  4 01:39:33 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 3 Sep 2012 14:39:33 -0700
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
Message-ID: <CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>

Please file a bug on JIRA for this. It seems like there's a bug in the
build system that prevented libtiff being built in your platform. For
the mean time, you can switch to the v1.2.stable branch if you really
need FS up and running right now.

On Mon, Sep 3, 2012 at 12:25 PM, Guillermo Ruiz Camauer
<grcamauer at gmail.com> wrote:
>
>
> Today, after issuing "make current", span_dsp wouldn't compile.  Since I had
> problems with this before, I issued "make spandsp-reconf" which fixed the
> problem last time, but not this time.
> While researching into the matter, I tried "git checkout --
> libs/spandsp/INSTALL" - followed by "make current" with no joy.  I also
> tried "git pull && ./bootstrap.sh && ./configure && make" to no avail.
> All of these where suggestions found in other threads of this list.
> I also tried "make sure".  After these attempts, I am getting the following
> error upon freeswitch startup:
>
> 2012-09-04 15:52:51.286376 [CRIT] switch_loadable_module.c:1310 Error
> Loading module /usr/local/freeswitch/mod/mod_spandsp.so
> **libtiff.so.5: cannot open shared object file: No such file or directory**.
>
> This is on:
> CentOS 6.3 Final
> Linux Freeswitch 2.6.32-279.5.2.el6.x86_64 #1 SMP Fri Aug 24 01:07:11 UTC
> 2012 x86_64 x86_64 x86_64 GNU/Linux
> FreeSWITCH Version 1.3.0+git~20120903T183548Z~f012b7abf8 (1.3.0; git at
> commit f012b7abf8 on Mon, 03 Sep 2012 18:35:48 Z) Started.
> Any tips will be appreciated.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Tue Sep  4 02:33:43 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Mon, 3 Sep 2012 19:33:43 -0300
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
Message-ID: <CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>

Thank you for your reply.

Just to add some more information, I tried "tiff-reconf" as per
http://wiki.freeswitch.org/wiki/Mod_spandsp and get the following error:

[root at Freeswitch freeswitch]# make tiff-reconf
cd libs/tiff-4.0.2 && autoreconf -fi
libtoolize: putting auxiliary files in AC_CONFIG_AUX_DIR, `config'.
libtoolize: copying file `config/ltmain.sh'
libtoolize: putting macros in AC_CONFIG_MACRO_DIR, `m4'.
libtoolize: copying file `m4/libtool.m4'
libtoolize: copying file `m4/ltoptions.m4'
libtoolize: copying file `m4/ltsugar.m4'
libtoolize: copying file `m4/ltversion.m4'
libtoolize: copying file `m4/lt~obsolete.m4'
cd libs/tiff-4.0.2 && sh ./configure.gnu
sh: ./configure.gnu: No such file or directory
make: *** [tiff-reconf] Error 127
[root at Freeswitch freeswitch]#

Any ideas?

Because I tried "git checkout -- libs/spandsp/INSTALL" git is not letting
me revert to stable:

[root at Freeswitch freeswitch]# git checkout v1.2.stable
error: You have local changes to 'libs/spandsp/INSTALL'; cannot switch
branches.
Any idea on how to reverse "git checkout -- libs/spandsp/INSTALL"?

Thanks in advance.


On Mon, Sep 3, 2012 at 6:39 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> Please file a bug on JIRA for this. It seems like there's a bug in the
> build system that prevented libtiff being built in your platform. For
> the mean time, you can switch to the v1.2.stable branch if you really
> need FS up and running right now.
>
> On Mon, Sep 3, 2012 at 12:25 PM, Guillermo Ruiz Camauer
> <grcamauer at gmail.com> wrote:
> >
> >
> > Today, after issuing "make current", span_dsp wouldn't compile.  Since I
> had
> > problems with this before, I issued "make spandsp-reconf" which fixed the
> > problem last time, but not this time.
> > While researching into the matter, I tried "git checkout --
> > libs/spandsp/INSTALL" - followed by "make current" with no joy.  I also
> > tried "git pull && ./bootstrap.sh && ./configure && make" to no avail.
> > All of these where suggestions found in other threads of this list.
> > I also tried "make sure".  After these attempts, I am getting the
> following
> > error upon freeswitch startup:
> >
> > 2012-09-04 15:52:51.286376 [CRIT] switch_loadable_module.c:1310 Error
> > Loading module /usr/local/freeswitch/mod/mod_spandsp.so
> > **libtiff.so.5: cannot open shared object file: No such file or
> directory**.
> >
> > This is on:
> > CentOS 6.3 Final
> > Linux Freeswitch 2.6.32-279.5.2.el6.x86_64 #1 SMP Fri Aug 24 01:07:11 UTC
> > 2012 x86_64 x86_64 x86_64 GNU/Linux
> > FreeSWITCH Version 1.3.0+git~20120903T183548Z~f012b7abf8 (1.3.0; git at
> > commit f012b7abf8 on Mon, 03 Sep 2012 18:35:48 Z) Started.
> > Any tips will be appreciated.
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bdfoster at endigotech.com  Tue Sep  4 03:08:35 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 3 Sep 2012 19:08:35 -0400
Subject: [Freeswitch-users] Reload dingalings jingle_profiles
In-Reply-To: <CAAQCs32yHMPcBOYoBwtq-db7pCFoqSC94tJPRubq_gC8GzQ4ow@mail.gmail.com>
References: <CAAQCs32yHMPcBOYoBwtq-db7pCFoqSC94tJPRubq_gC8GzQ4ow@mail.gmail.com>
Message-ID: <CAKV-jFwOhjn6wXqYDSE7-fD7ndnUdoUmkdshRaOF06sZYxQC9w@mail.gmail.com>

reload mod_dingaling

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 3, 2012 4:56 PM, "Nikolay Rogoshchenkov" <nickolayr at gmail.com> wrote:

>
> Hello,
> Does anybody know is it possible to reload *dingalings jingle_profiles*from console?
>
> PS: *reloadxml *and *dingalings reload* did not helps
>
> Thanks.
> --
> Rogoshchenkov Nikolay
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Sep  4 04:53:04 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 3 Sep 2012 17:53:04 -0700
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
Message-ID: <CAMWiMy_7U4-91t33gRDdZ5e-+D_G+owMiD=4QBpTD3Ovr50+XQ@mail.gmail.com>

To switch to the stable branch what you'll have to do first is run a
git clean -f -d -x. Then, run a git reset --hard, after that
re-attempt the switch from master (testing) to v1.2.stable

On Mon, Sep 3, 2012 at 3:33 PM, Guillermo Ruiz Camauer
<grcamauer at gmail.com> wrote:
> Thank you for your reply.
>
> Just to add some more information, I tried "tiff-reconf" as per
> http://wiki.freeswitch.org/wiki/Mod_spandsp and get the following error:
>
> [root at Freeswitch freeswitch]# make tiff-reconf
> cd libs/tiff-4.0.2 && autoreconf -fi
> libtoolize: putting auxiliary files in AC_CONFIG_AUX_DIR, `config'.
> libtoolize: copying file `config/ltmain.sh'
> libtoolize: putting macros in AC_CONFIG_MACRO_DIR, `m4'.
> libtoolize: copying file `m4/libtool.m4'
> libtoolize: copying file `m4/ltoptions.m4'
> libtoolize: copying file `m4/ltsugar.m4'
> libtoolize: copying file `m4/ltversion.m4'
> libtoolize: copying file `m4/lt~obsolete.m4'
> cd libs/tiff-4.0.2 && sh ./configure.gnu
> sh: ./configure.gnu: No such file or directory
> make: *** [tiff-reconf] Error 127
> [root at Freeswitch freeswitch]#
>
> Any ideas?
>
> Because I tried "git checkout -- libs/spandsp/INSTALL" git is not letting me
> revert to stable:
>
> [root at Freeswitch freeswitch]# git checkout v1.2.stable
> error: You have local changes to 'libs/spandsp/INSTALL'; cannot switch
> branches.
> Any idea on how to reverse "git checkout -- libs/spandsp/INSTALL"?
>
> Thanks in advance.
>
>
> On Mon, Sep 3, 2012 at 6:39 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> Please file a bug on JIRA for this. It seems like there's a bug in the
>> build system that prevented libtiff being built in your platform. For
>> the mean time, you can switch to the v1.2.stable branch if you really
>> need FS up and running right now.
>>
>> On Mon, Sep 3, 2012 at 12:25 PM, Guillermo Ruiz Camauer
>> <grcamauer at gmail.com> wrote:
>> >
>> >
>> > Today, after issuing "make current", span_dsp wouldn't compile.  Since I
>> > had
>> > problems with this before, I issued "make spandsp-reconf" which fixed
>> > the
>> > problem last time, but not this time.
>> > While researching into the matter, I tried "git checkout --
>> > libs/spandsp/INSTALL" - followed by "make current" with no joy.  I also
>> > tried "git pull && ./bootstrap.sh && ./configure && make" to no avail.
>> > All of these where suggestions found in other threads of this list.
>> > I also tried "make sure".  After these attempts, I am getting the
>> > following
>> > error upon freeswitch startup:
>> >
>> > 2012-09-04 15:52:51.286376 [CRIT] switch_loadable_module.c:1310 Error
>> > Loading module /usr/local/freeswitch/mod/mod_spandsp.so
>> > **libtiff.so.5: cannot open shared object file: No such file or
>> > directory**.
>> >
>> > This is on:
>> > CentOS 6.3 Final
>> > Linux Freeswitch 2.6.32-279.5.2.el6.x86_64 #1 SMP Fri Aug 24 01:07:11
>> > UTC
>> > 2012 x86_64 x86_64 x86_64 GNU/Linux
>> > FreeSWITCH Version 1.3.0+git~20120903T183548Z~f012b7abf8 (1.3.0; git at
>> > commit f012b7abf8 on Mon, 03 Sep 2012 18:35:48 Z) Started.
>> > Any tips will be appreciated.
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>


From khuenm at vega.com.vn  Tue Sep  4 05:49:44 2012
From: khuenm at vega.com.vn (Khue Nguyen Minh)
Date: Tue, 4 Sep 2012 08:49:44 +0700
Subject: [Freeswitch-users] No Dialplan on answer channel
In-Reply-To: <CALNZuwnKiXKR6S4eqiuOizu=1C+rBuGpxPnrzHDnU3cu9RvB=w@mail.gmail.com>
References: <CAO-yT6hxe8eCcuT36WEjqmLDKYYfyGHnv3c2O_Opz3q6UMOhmA@mail.gmail.com>
	<CAKzWOxXyYZH8UUSDEVzvcF+avzFLYQaT-6sNbvsozSiW-FmsJQ@mail.gmail.com>
	<CALNZuwnKiXKR6S4eqiuOizu=1C+rBuGpxPnrzHDnU3cu9RvB=w@mail.gmail.com>
Message-ID: <CAO-yT6gO4ymWJsM14H6EmSV0Vdc1AVJueTvABN0MOXkfe72gjw@mail.gmail.com>

Hi Avi,

My javascript work normal (I tested it in a xml dialplan). I still try with
originate sofia/external/1000 at somewhere.com &javascript('hello.js
/opt/freeswitch/sounds/2.mp3') but It doesn't work.

Brs,
Khue Nguyen.

2012/8/31 Avi Marcus <avi at avimarcus.net>

> 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
> McAfee SiteAdvisor Warning
> This e-mail message contains potentially unsafe links to these sites:
> ostag.org [image: more info...]<https://www.siteadvisor.com/sites/http://www.ostag.org/-?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.229&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
> freeswitchsolutions.com [image: more info...]<https://www.siteadvisor.com/sites/http://www.freeswitchsolutions.com/-?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.229&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>
>
> 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.organd 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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Sep  4 07:13:56 2012
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Mon, 3 Sep 2012 20:13:56 -0700 (PDT)
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
Message-ID: <1346728436032-7582519.post@n2.nabble.com>

If I undestand it correctly, the /configure.gnu/ file isn't native to any
libtiff package. Probably, it was added by FS developers to produce
non-shared (static) libraries. Unfortunately, when libtiff package got
upgraded to v4.0.2, this file which is needed by the FS main Makefile.am
file isn't found anywhere in the libtiff v4.0.2 package. It is very simple
to fix this problem, i.e. replace the /./configure.gnu/ line in the
Makefile.am file with something like /./configure --disable-shared
--with-pic/. The old libs/tiff-3.8.2/configure.gnu is a simple shell script
file and it contains only three lines as shown below:



Guillermo Ruiz Camauer wrote
> 
> Thank you for your reply.
> 
> Just to add some more information, I tried "tiff-reconf" as per
> http://wiki.freeswitch.org/wiki/Mod_spandsp and get the following error:
> 
> [root at Freeswitch freeswitch]# make tiff-reconf
> cd libs/tiff-4.0.2 && autoreconf -fi
> libtoolize: putting auxiliary files in AC_CONFIG_AUX_DIR, `config'.
> libtoolize: copying file `config/ltmain.sh'
> libtoolize: putting macros in AC_CONFIG_MACRO_DIR, `m4'.
> libtoolize: copying file `m4/libtool.m4'
> libtoolize: copying file `m4/ltoptions.m4'
> libtoolize: copying file `m4/ltsugar.m4'
> libtoolize: copying file `m4/ltversion.m4'
> libtoolize: copying file `m4/lt~obsolete.m4'
> cd libs/tiff-4.0.2 && sh ./configure.gnu
> sh: ./configure.gnu: No such file or directory
> make: *** [tiff-reconf] Error 127
> [root at Freeswitch freeswitch]#
> 
> Any ideas?



-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Missing-libtiff-after-make-current-tp7582512p7582519.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Tue Sep  4 07:40:34 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 3 Sep 2012 20:40:34 -0700 (PDT)
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <1346728436032-7582519.post@n2.nabble.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
Message-ID: <1346730034461-7582520.post@n2.nabble.com>

I re-added the missing configure.gnu file.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Missing-libtiff-after-make-current-tp7582512p7582520.html
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From philq at qsystemsengineering.com  Tue Sep  4 03:07:13 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Mon, 03 Sep 2012 19:07:13 -0400
Subject: [Freeswitch-users] SpanDSP compile failure with latest Git
Message-ID: <006901cd8a28$e14b5d30$a3e21790$@com>

Tried ./bootstrap.sh, ./configure, and spandsp-reconf to no avail.

 

Make yields the following error:




/usr/src/freeswitch/libs/spandsp/src/spandsp/t42.h:67: error: expected
declaration specifiers or ?...? before ?tsize_t?

/usr/src/freeswitch/libs/spandsp/src/spandsp/t42.h:69: error: expected
declaration specifiers or ?...? before ?tsize_t?

/usr/src/freeswitch/libs/spandsp/src/spandsp/t42.h:69: error: expected
declaration specifiers or ?...? before ?tsize_t?

make[1]: *** [freeswitch-switch.o] Error 1

make: *** [all] Error 2




 

Spandsp-reconf yields the following error:




In file included from fax.c:80:

spandsp/t42.h:61: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:61: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:63: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:63: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:65: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:65: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:67: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:67: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:69: error: expected declaration specifiers or '...' before
'tsize_t'

spandsp/t42.h:69: error: expected declaration specifiers or '...' before
'tsize_t'

make[3]: *** [fax.lo] Error 1

make[3]: Leaving directory `/usr/src/freeswitch/libs/spandsp/src'

make[2]: *** [all] Error 2

make[2]: Leaving directory `/usr/src/freeswitch/libs/spandsp/src'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/src/freeswitch/libs/spandsp'

make: *** [spandsp-reconf] Error 2




 

Phil Quesinberry

Q Systems Engineering, Inc.

Electronic Controls and Embedded Systems Development

(410) 969-8002

http://www.qsystemsengineering.com

 

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From curriegrad2004 at gmail.com  Tue Sep  4 09:03:03 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 3 Sep 2012 22:03:03 -0700
Subject: [Freeswitch-users] SpanDSP compile failure with latest Git
In-Reply-To: <006901cd8a28$e14b5d30$a3e21790$@com>
References: <006901cd8a28$e14b5d30$a3e21790$@com>
Message-ID: <CAMWiMy8vHXjkoq52NXGXe+=0SC=_bTf+KJON3JiEnhC9xDVvQg@mail.gmail.com>

Are you using the master branch or the stable "v1.2.stable" branch?

If you are going to compile the master branch for a production system,
then I would strongly recommend that you run a git clean first, reset
--hard and then switch over the v1.2.stable branch. The master branch
is now more or less a "playground" on developers so they can do what
they need to do to further enhance FreeSWITCH without having to worry
about breaking other things.

On Mon, Sep 3, 2012 at 4:07 PM, Phil Quesinberry
<philq at qsystemsengineering.com> wrote:
> Tried ./bootstrap.sh, ./configure, and spandsp-reconf to no avail.
>
>
>
> Make yields the following error:
>
> ?
>
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t42.h:67: error: expected
> declaration specifiers or ?...? before ?tsize_t?
>
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t42.h:69: error: expected
> declaration specifiers or ?...? before ?tsize_t?
>
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t42.h:69: error: expected
> declaration specifiers or ?...? before ?tsize_t?
>
> make[1]: *** [freeswitch-switch.o] Error 1
>
> make: *** [all] Error 2
>
> ?
>
>
>
> Spandsp-reconf yields the following error:
>
> ?
>
> In file included from fax.c:80:
>
> spandsp/t42.h:61: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:61: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:63: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:63: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:65: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:65: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:67: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:67: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:69: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> spandsp/t42.h:69: error: expected declaration specifiers or '...' before
> 'tsize_t'
>
> make[3]: *** [fax.lo] Error 1
>
> make[3]: Leaving directory `/usr/src/freeswitch/libs/spandsp/src'
>
> make[2]: *** [all] Error 2
>
> make[2]: Leaving directory `/usr/src/freeswitch/libs/spandsp/src'
>
> make[1]: *** [all-recursive] Error 1
>
> make[1]: Leaving directory `/usr/src/freeswitch/libs/spandsp'
>
> make: *** [spandsp-reconf] Error 2
>
> ?
>
>
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Tue Sep  4 13:27:15 2012
From: govoiper at gmail.com (SamyGo)
Date: Tue, 4 Sep 2012 14:27:15 +0500
Subject: [Freeswitch-users] Limitation
In-Reply-To: <CAMwZ-tJdeDXKt3qEBZaqSSgkfZBqKS0+bDwugewqLGrkyc_5mg@mail.gmail.com>
References: <502B8B67.3090306@pripojtese.net>
	<CAMwZ-tJdeDXKt3qEBZaqSSgkfZBqKS0+bDwugewqLGrkyc_5mg@mail.gmail.com>
Message-ID: <CAJUJwtgda662nZQgx6p7Nyjq_JmzHuCwMnuxRWWv5OMe=8VnFA@mail.gmail.com>

Hi,

" Usage for CONCURRENT_CALLS_IN_FREE_MODE_calls_max is now 0 " - Thats
definitely not FreeSwitch. just find this particular variable in vbilling
or fusionpbx code and try increasing the decrementing counter.

Thanks,
Sammy



On Mon, Sep 3, 2012 at 12:59 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Wed, Aug 15, 2012 at 5:43 AM, Jakub Tencl <info at pripojtese.net> wrote:
> > 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?
>
> Wouldn't the limitation be on the vBilling side (if there is any)? It
> doesn't seem like this list is the right place to ask. Anyway, good
> luck and let us know what you learn.
>
>
> 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 bdfoster at endigotech.com  Tue Sep  4 14:04:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 4 Sep 2012 06:04:27 -0400
Subject: [Freeswitch-users] Limitation
In-Reply-To: <CAJUJwtgda662nZQgx6p7Nyjq_JmzHuCwMnuxRWWv5OMe=8VnFA@mail.gmail.com>
References: <502B8B67.3090306@pripojtese.net>
	<CAMwZ-tJdeDXKt3qEBZaqSSgkfZBqKS0+bDwugewqLGrkyc_5mg@mail.gmail.com>
	<CAJUJwtgda662nZQgx6p7Nyjq_JmzHuCwMnuxRWWv5OMe=8VnFA@mail.gmail.com>
Message-ID: <CAKV-jFwcDit0hBz6Fhv9V9JWY78HxKERveWx1ZZXivbpNG_p3Q@mail.gmail.com>

It is a limitation imposed by vbilling. However it isn't one you are
supposed to be able to change (easily). The author has decided to impose
this limitation for his own reasons, and my intent is to not speak for him
here.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 4, 2012 5:29 AM, "SamyGo" <govoiper at gmail.com> wrote:

> Hi,
>
> " Usage for CONCURRENT_CALLS_IN_FREE_MODE_calls_max is now 0 " - Thats
> definitely not FreeSwitch. just find this particular variable in vbilling
> or fusionpbx code and try increasing the decrementing counter.
>
> Thanks,
> Sammy
>
>
>
> On Mon, Sep 3, 2012 at 12:59 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Wed, Aug 15, 2012 at 5:43 AM, Jakub Tencl <info at pripojtese.net> wrote:
>> > 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?
>>
>> Wouldn't the limitation be on the vBilling side (if there is any)? It
>> doesn't seem like this list is the right place to ask. Anyway, good
>> luck and let us know what you learn.
>>
>>
>> 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
>
>
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From Vladislav.Grishin at vts24.ru  Tue Sep  4 14:36:32 2012
From: Vladislav.Grishin at vts24.ru (=?KOI8-R?Q?=22=E7=D2=C9=DB=C9=CE_=F7=2E=F3=2E=22?=)
Date: Tue, 04 Sep 2012 14:36:32 +0400
Subject: [Freeswitch-users] How to compile and install ESL module for python
	2.6 on CentOS5.7?
In-Reply-To: <4F6C8271.7020800@vts24.ru>
References: <4F6C8271.7020800@vts24.ru>
Message-ID: <5045D9B0.2000205@vts24.ru>

CentOS5.7 include python-2.4.3 by default

I install python 2.6 from sources by default into /usr/local/bin/ and 
change a strings in

/usr/src/freeswitch.git/libs/esl/python/Makefile

# more /usr/src/freeswitch.git/libs/esl/python/Makefile
LOCAL_CFLAGS=`/usr/local/bin/python ./python-config --includes`
LOCAL_LDFLAGS=`/usr/local/bin/python ./python-config --ldflags`
SITE_DIR=$(DESTDIR)/`/usr/local/bin/python -c "from distutils.sysconfig 
import get_python_lib; print get_python_lib()"`

and

#more /usr/src/freeswitch.git/libs/esl/python/python-config

#!/usr/local/bin/python

SWIG 2.0.8 is also installed into /usr/local/bin/  with configure 
options --with-python=/usr/local/bin/python --without-python3 --without-pcre


run "make pymod" into /usr/src/freeswitch.git/libs/esl directory

Output is

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 python
make[1]: Entering directory `/usr/src/freeswitch.git/libs/esl/python'
g++  -I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g 
-ggdb -I../../libs/libedit/src/ -fPIC `/usr/local/bin/python 
./python-config --includes` -c esl_wrap.cpp -o esl_wrap.o
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a `/usr/local/bin/python 
./python-config --ldflags` -o _ESL.so -L.
/usr/bin/ld: /usr/local/lib/python2.6/config/libpython2.6.a(abstract.o): 
relocation R_X86_64_32 against `a local symbol' can not be used when 
making a shared object; recompile with -fPIC
/usr/local/lib/python2.6/config/libpython2.6.a: could not read symbols: 
Bad value
collect2: ld returned 1 exit status
make[1]: *** [_ESL.so] Error 1
make[1]: Leaving directory `/usr/src/freeswitch.git/libs/esl/python'
make: *** [pymod] Error 2

# /usr/bin/ld /usr/local/lib/python2.6/config/libpython2.6.a
/usr/bin/ld: warning: cannot find entry symbol _start; not setting start 
address

# file /usr/local/lib/python2.6/config/libpython2.6.a
/usr/local/lib/python2.6/config/libpython2.6.a: current ar archive



in what there can be a problem?



Vladislav Grishin

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From coopara at go2.pl  Tue Sep  4 14:47:13 2012
From: coopara at go2.pl (=?UTF-8?Q?coopara?=)
Date: Tue, 04 Sep 2012 12:47:13 +0200
Subject: [Freeswitch-users] =?utf-8?q?Fax_through_T=2E38_or_G=2E711?=
Message-ID: <460ae07a.27a7a891.5045dc31.7edfa@go2.pl>

Hi!
?
I would like to control which codec FS is using to sending faxes. Under certain circumstances, I would only use G.711 and in the other situations only T.38. How can I control it in the dialplan?
?
Thanks for any help.
?
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From freeswitch-list at puzzled.xs4all.nl  Tue Sep  4 16:06:34 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 04 Sep 2012 14:06:34 +0200
Subject: [Freeswitch-users] How to compile and install ESL module for
 python 2.6 on CentOS5.7?
In-Reply-To: <5045D9B0.2000205@vts24.ru>
References: <4F6C8271.7020800@vts24.ru> <5045D9B0.2000205@vts24.ru>
Message-ID: <5045EECA.2040103@puzzled.xs4all.nl>

On 04-09-12 12:36, "?????? ?.?." wrote:
> /usr/bin/ld: /usr/local/lib/python2.6/config/libpython2.6.a(abstract.o):
> relocation R_X86_64_32 against `a local symbol' can not be used when
> making a shared object; recompile with -fPIC

Did you see this error? ^^^ It clearly states that something was 
compiled without -fPIC. Try adding it.

Regards,
Patrick



From jaybinks at gmail.com  Tue Sep  4 16:43:54 2012
From: jaybinks at gmail.com (jay binks)
Date: Tue, 4 Sep 2012 22:43:54 +1000
Subject: [Freeswitch-users] Fax through T.38 or G.711
In-Reply-To: <460ae07a.27a7a891.5045dc31.7edfa@go2.pl>
References: <460ae07a.27a7a891.5045dc31.7edfa@go2.pl>
Message-ID: <CAO47HpcB-i7m=Vk34n3aToxDxzzkyF76WEtvACi2f9HQGBx=ng@mail.gmail.com>

http://lmgtfy.com/?q=freeswitch+enable+t38&l=1


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


we dont mind helping, but sometimes... seriously
google is faster for simple stuff like this.


On 4 September 2012 20:47, coopara <coopara at go2.pl> wrote:

> Hi!
>
> I would like to control which codec FS is using to sending faxes. Under
> certain circumstances, I would only use G.711 and in the other situations
> only T.38. How can I control it in the dialplan?
>
> Thanks for any help.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 amit.pureenergy at gmail.com  Tue Sep  4 17:13:51 2012
From: amit.pureenergy at gmail.com (Amit Sethi)
Date: Tue, 4 Sep 2012 18:43:51 +0530
Subject: [Freeswitch-users] How to compile and install ESL module for
 python 2.6 on CentOS5.7?
In-Reply-To: <5045D9B0.2000205@vts24.ru>
References: <4F6C8271.7020800@vts24.ru> <5045D9B0.2000205@vts24.ru>
Message-ID: <CAN+DPADPZnJS1Pu5uVWiir_OMxtG2Oej3H1mK8kWws6m4iPK7w@mail.gmail.com>

> /usr/bin/ld: /usr/local/lib/python2.6/config/libpython2.6.a(abstract.o):
> relocation R_X86_64_32 against `a local symbol' can not be used when making
> a shared object; recompile with -fPIC
> /usr/local/lib/python2.6/config/libpython2.6.a: could not read symbols: Bad
> value

Are you sure you have python dev packages installed . I think this has
to do missing symbols for python lib which swig might need.



Thanks
Amit

-- 
A-M-I-T S|S


From Vladislav.Grishin at vts24.ru  Tue Sep  4 17:29:32 2012
From: Vladislav.Grishin at vts24.ru (=?UTF-8?B?ItCT0YDQuNGI0LjQvSDQki7QoS4i?=)
Date: Tue, 04 Sep 2012 17:29:32 +0400
Subject: [Freeswitch-users] [resolved]Re: How to compile and install ESL
 module for python 2.6 on CentOS5.7?
In-Reply-To: <5045EECA.2040103@puzzled.xs4all.nl>
References: <4F6C8271.7020800@vts24.ru> <5045D9B0.2000205@vts24.ru>
	<5045EECA.2040103@puzzled.xs4all.nl>
Message-ID: <5046023C.4000903@vts24.ru>

all ok,  after I add "-fPIC"  into /usr/src/Python-2.6.8/Makefile
...
CFLAGS=         $(BASECFLAGS) -g -O2 $(OPT) $(EXTRA_CFLAGS) -fPIC
...


Thanks,
Vladislav Grishin


From Nabble_01394 at slickdeals.endjunk.com  Tue Sep  4 17:56:46 2012
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Tue, 4 Sep 2012 06:56:46 -0700 (PDT)
Subject: [Freeswitch-users] SpanDSP compile failure with latest Git
In-Reply-To: <006901cd8a28$e14b5d30$a3e21790$@com>
References: <006901cd8a28$e14b5d30$a3e21790$@com>
Message-ID: <1346767006119-7582531.post@n2.nabble.com>

Yup. This has been the issue since spandsp started introducing some
meaningful TIFF/FX stuff in git commit
cbb28e2ae061e8615dde948e17d66c4d6178b142.



-----
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/SpanDSP-compile-failure-with-latest-Git-tp7582521p7582531.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Hector.Geraldino at ipsoft.com  Tue Sep  4 18:29:08 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Tue, 4 Sep 2012 10:29:08 -0400
Subject: [Freeswitch-users] Query about Valet Parking
In-Reply-To: <006001cd8992$d82fbc50$888f34f0$@co.in>
References: <CALvF6vAoMBhg0N_jhiEeUFRxHdB2Saiv5iPf=uFNBmWDvxR5XQ@mail.gmail.com>
	<01ed01cd8790$0800e850$1802b8f0$@co.in>
	<006001cd8992$d82fbc50$888f34f0$@co.in>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD073F6F3054@NY1-EXMB-01.ip-soft.net>

You can transfer the call to an extension (using the transfer app in the dialplan) which is controlled by your java application using Java ESL in outbound mode. Once you have control of the call, you can check all the call variables by looking at the call headers.

Here's the flow:
 dialplan (ivr) -> user's select option -> save option in variable -> transfer to extension 1234 -> java app

I recommend you to use the Java ESL client (http://wiki.freeswitch.org/wiki/Java_ESL_Client), check the test examples and go from there.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin Tomer
Sent: Monday, September 03, 2012 1:13 AM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] Query about Valet Parking

Hi,

I am building an application, in which callers will call a designated number, will be presented with an IVR menu and then their call will be parked at an extension. I need to call a Java API after that and pass on Customer's IVR selection to that API. How can I do that? Any pointers?

Thanks

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 lists at kavun.ch  Tue Sep  4 18:35:15 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 4 Sep 2012 10:35:15 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
Message-ID: <152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>

Hi MC,

Thanks a bunch for your reply, sorry for the delay.
Here are my logs:
http://pastebin.freeswitch.org/19831

Any idea would be greatly appreciated.

Best,
Emrah


On Aug 28, 2012, at 10:56 AM, Michael Collins <msc at freeswitch.org> wrote:

> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From cjbujold at accra.ca  Tue Sep  4 19:40:25 2012
From: cjbujold at accra.ca (Charles Bujold)
Date: Tue, 4 Sep 2012 12:40:25 -0300
Subject: [Freeswitch-users] routing error????
Message-ID: <003801cd8ab3$9a255790$ce7006b0$@accra.ca>

This morning I am encountering the following error:

 

The originating call is from extension 250  at 192.168.20.23  to extension
425 at 192.168.25.13 ( same server two different office locations).  from
what I see I think it is some routing issue but not certain.  How can I
determine what is the cause? And how can I fix it?

 

Thanks

cjb

 

 

2012-09-04 12:17:44.001046 [DEBUG] switch_core_session.c:1229 Send signal
sofia/internal/sip:425 at 192.168.25.13:5062 [BREAK]

2012-09-04 12:17:44.001046 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/sip:425 at 192.168.25.13:5062) Running State Change CS_HANGUP

2012-09-04 12:17:44.001046 [NOTICE] switch_ivr_originate.c:2544 Cannot
create outgoing channel of type [user] cause: [NO_ANSWER]

2012-09-04 12:17:44.001046 [DEBUG] switch_ivr_originate.c:3458 Originate
Resulted in Error Cause: 19 [NO_ANSWER]

2012-09-04 12:17:44.001046 [INFO] mod_dptools.c:3027 Originate Failed.
Cause: NO_ANSWER

 

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From msc at freeswitch.org  Tue Sep  4 20:46:47 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Sep 2012 09:46:47 -0700
Subject: [Freeswitch-users] No Dialplan on answer channel
In-Reply-To: <CAO-yT6gO4ymWJsM14H6EmSV0Vdc1AVJueTvABN0MOXkfe72gjw@mail.gmail.com>
References: <CAO-yT6hxe8eCcuT36WEjqmLDKYYfyGHnv3c2O_Opz3q6UMOhmA@mail.gmail.com>
	<CAKzWOxXyYZH8UUSDEVzvcF+avzFLYQaT-6sNbvsozSiW-FmsJQ@mail.gmail.com>
	<CALNZuwnKiXKR6S4eqiuOizu=1C+rBuGpxPnrzHDnU3cu9RvB=w@mail.gmail.com>
	<CAO-yT6gO4ymWJsM14H6EmSV0Vdc1AVJueTvABN0MOXkfe72gjw@mail.gmail.com>
Message-ID: <CAKzWOxV1VUz2xGe50An9oWT53OoY0pkx_0Nf4QVLCeV-aSt1yw@mail.gmail.com>

"It doesn't work" is a big vague. Could you be more specific?
-MC

On Mon, Sep 3, 2012 at 6:49 PM, Khue Nguyen Minh <khuenm at vega.com.vn> wrote:

> Hi Avi,
>
> My javascript work normal (I tested it in a xml dialplan). I still try
> with originate sofia/external/1000 at somewhere.com &javascript('hello.js
> /opt/freeswitch/sounds/2.mp3') but It doesn't work.
>
> Brs,
> Khue Nguyen.
>
> 2012/8/31 Avi Marcus <avi at avimarcus.net>
>
>> 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
>>  McAfee SiteAdvisor Warning
>> This e-mail message contains potentially unsafe links to these sites:
>> ostag.org [image: more info...]<https://www.siteadvisor.com/sites/http://www.ostag.org/-?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.229&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>> freeswitchsolutions.com [image: more info...]<https://www.siteadvisor.com/sites/http://www.freeswitchsolutions.com/-?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.229&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>>
>>
>> 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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mailing-lists at phoenixinternet.net  Tue Sep  4 20:48:49 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Tue, 04 Sep 2012 09:48:49 -0700
Subject: [Freeswitch-users] sofia xmlstatus profile internal reg
In-Reply-To: <BLU170-DS18E69BCC0DBB9E8222986FD7A50@phx.gbl>
References: <50414E03.40800@phoenixinternet.net>
	<16a801cd87fb$ccc73200$66559600$@freeswitch.org>
	<BLU170-DS18E69BCC0DBB9E8222986FD7A50@phx.gbl>
Message-ID: <504630F1.40502@phoenixinternet.net>

I attached my notes into the comment of your ticket since it's title 
encompasses my issue as well. Thanks for bringing that to my attention. 
If you are running FusionPBX, I have written some php code that could be 
modified to fix your issue. It will cause some issues though once they 
fix the bug. I posted it in the FusionPBX bug tracker as issue 303.

http://code.google.com/p/fusionpbx/issues/list

Gilbert

On 8/31/2012 11:28 PM, Mr Nathan Downes wrote:
>
> Similar to http://jira.freeswitch.org/browse/FS-3971
>
> Had same problem with & in the contact field
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Michael Collins
> *Sent:* Saturday, 1 September 2012 1:47 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sofia xmlstatus profile internal reg
>
> Definitely put this on jira.freeswitch.org <http://jira.freeswitch.org>.
> -MC
>
> On Friday, August 31, 2012, Gilbert T. Gutierrez, Jr. 
> <mailing-lists at phoenixinternet.net 
> <mailto:mailing-lists at phoenixinternet.net>> wrote:
> > I believe I have found a bug in the response of "sofia xmlstatus profile
> > internal reg". I just do not know how to report it.
> >
> > It is not returning all special characters in UTF-8 format. For example
> > '<' should be returned as &lt;. In the <contact> xml tag it returns a
> > correct string. In the <agent> xml tag it does not return it correctly.
> > Below is an example from my test server. The 3rd entry has an agent that
> > returns both less than and greater than symbols and freeswitch is not
> > sending them as &lt; or &gt;.
> >
> > freeswitch at internal> sofia xmlstatus profile internal reg
> > <?xml version="1.0" encoding="ISO-8859-1"?>
> > <profile>
> >    <registrations>
> >      <registration>
> > <call-id>000ed700-06320003-0572ebcb-59776109 at 172.16.5.83 
> <mailto:000ed700-06320003-0572ebcb-59776109 at 172.16.5.83></call-id>
> >          <user>190 at 192.168.1.6 <mailto:190 at 192.168.1.6></user>
> >          <contact>"user"
> > &lt;sip:190 at 192.168.1.90:1307;transport=udp&gt;</contact>
> >          <agent>Cisco-CP7940G/8.0</agent>
> >          <status>Registered(UDP)(unknown) exp(2012-08-31 16:42:28)
> > expsecs(153)</status>
> >          <host>test.abc.com <http://test.abc.com></host>
> >  <network-ip>192.168.1.90</network-ip>
> >          <network-port>1307</network-port>
> >          <sip-auth-user>190</sip-auth-user>
> >          <sip-auth-realm>test.abc.com 
> <http://test.abc.com></sip-auth-realm>
> >          <mwi-account>190 at 192.168.1.6 
> <mailto:190 at 192.168.1.6></mwi-account>
> >      </registration>
> >      <registration>
> > <call-id>NTVkZWI5NDM5Yzk0ZWY4M2Q0NDcyNDE0YWM0OTJkMWE.</call-id>
> >          <user>106 at 192.168.1.6 <mailto:106 at 192.168.1.6></user>
> >          <contact>"Gilbert Gutierrez"
> > &lt;sip:106 at 192.168.1.90:18360;rinstance=4645682e9800585c&gt;</contact>
> >          <agent>X-Lite release 5.0.0 stamp 67284</agent>
> >          <status>Registered(UDP)(unknown) exp(2012-08-31 17:06:16)
> > expsecs(1581)</status>
> >          <host>test.abc.com <http://test.abc.com></host>
> >  <network-ip>192.168.1.90</network-ip>
> >          <network-port>18360</network-port>
> >          <sip-auth-user>106</sip-auth-user>
> >          <sip-auth-realm>test.abc.com 
> <http://test.abc.com></sip-auth-realm>
> >          <mwi-account>106 at 192.168.1.6 
> <mailto:106 at 192.168.1.6></mwi-account>
> >      </registration>
> >      <registration>
> > <call-id>QC8jV0-xAG1nCDtN2 at test.abc.com 
> <mailto:QC8jV0-xAG1nCDtN2 at test.abc.com></call-id>
> >          <user>180 at 192.168.1.6 <mailto:180 at 192.168.1.6></user>
> >          <contact>PI FAX &lt;sip:180 at 192.168.1.114:5060&gt;</contact>
> >          <agent>Patton Smartlink 4020 <3.01.002 20 EN n0
> > (1214)><00a0ba017198></agent>
> >          <status>Registered(UDP)(unknown) exp(2012-08-31 17:24:31)
> > expsecs(2676)</status>
> >          <host>test.abc.com <http://test.abc.com></host>
> >  <network-ip>192.168.1.114</network-ip>
> >          <network-port>5060</network-port>
> >          <sip-auth-user>180</sip-auth-user>
> >          <sip-auth-realm>test.abc.com 
> <http://test.abc.com></sip-auth-realm>
> >          <mwi-account>180 at 192.168.1.6 
> <mailto:180 at 192.168.1.6></mwi-account>
> >      </registration>
> >    </registrations>
> > </profile>
> >
> >
> > Thank you,
> > Gilbert T. Gutierrez, Jr.
> > Operations Manager
> > Phoenix Internet
> >
> > 
> _________________________________________________________________________
> > 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 philq at qsystemsengineering.com  Tue Sep  4 19:33:41 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Tue, 4 Sep 2012 08:33:41 -0700 (PDT)
Subject: [Freeswitch-users] SpanDSP compile failure with latest Git
In-Reply-To: <1346767006119-7582531.post@n2.nabble.com>
References: <006901cd8a28$e14b5d30$a3e21790$@com>
	<1346767006119-7582531.post@n2.nabble.com>
Message-ID: <1346772821283-7582534.post@n2.nabble.com>

Thanks Curriegrad/Mazilo,

I was using the master branch but switched over to 1.2.stable last night
after running into this problem and it?s working well.  There was in issue
introduced in an earlier version of the master branch that I was running
where the switch would crash out if an extension tried to dial a code that
mapped to a SIP URI where the system was down.  I wanted to try the latest
build before filing a Jira.

While the master branch is surprisingly stable, I think I?ll go ahead and
stick to 1.2.stable for now since this is in fact a production system.  If
you live on the bleeding edge long enough, it will be your own blood that
you find there.

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SpanDSP-compile-failure-with-latest-Git-tp7582521p7582534.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue Sep  4 20:50:08 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Sep 2012 09:50:08 -0700
Subject: [Freeswitch-users] FreeSwitch SIP Registration
In-Reply-To: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
References: <CAFvodxUaBwu3_YjECGkvED3urSLhempTQ=AwkAB5E+eBbAo13g@mail.gmail.com>
Message-ID: <CAKzWOxV3B9hF3OCRMnBXNvFbPcXMoqgX1br+_63n28u7yMBQ5w@mail.gmail.com>

I take it your copies of the FreeSWITCH books have not yet arrived, because
this is covered in detail. ;)

If you are trying to get a phone connected to FreeSWITCH then I suggest you
read this older but still very relevant article:
http://www.linuxpromagazine.com/Issues/2009/106/FreeSWITCH

It's only a few pages and discusses (among other things) exactly what you
are trying to do w/ X-Lite.
-MC

On Mon, Sep 3, 2012 at 6:02 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi All,
>
>           I am trying to register  to freeswitch using X-lite i have done
> necessary configuration on X-lite when the X-lite tries to REGISTER to
> freeswitch it is saying 403 forbidden my sofia conf xml is like below  i
> followed the wiki page to create the sofia configuration file.I dont know
> the procedure that i followed to register the user is correct or not.Please
> suggest me how can i do the sip user registeration
> <profile name="My Name">
>  <gateways>
>  </gateways>
>  <aliases>
>  </aliases>
>  <domains>
>   <domain name="local_ip" alias="false" parse="true"/>
>   <settings>
>   <param name="challenge-realm" value="auto_from"/>
>   <param name="force-register-domain" value="local_ip"/>
>   <param name="force-subscription-domain" value="local_ip"/>
>   <param name="send-message-query-on-register" value="false"/>
>  </settings>
> </profile>
>
>
>
> 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jerry.richards at teotech.com  Tue Sep  4 20:53:35 2012
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 4 Sep 2012 16:53:35 +0000
Subject: [Freeswitch-users] Video Codec Sent In SDP Offer
Message-ID: <1545146083A72C4DB7B66584B7E5D9841D15BD5C@BY2PRD0410MB377.namprd04.prod.outlook.com>

If a call is made from a phone that does not include a video codec (i.e. H.264), Freeswitch still offers H.264 as a codec in the INVITE to the callee.  I think this is because H.264 is included in the inbound-codec-prefs and outbound-codec-prefs of the sip_profile (which is needed to support video calls).

Is there an easy way to configure Freeswitch to exclude H.264 based on the caller's INVITE?  Because it never would make sense for it to offer H.264 in this scenario.

Best Regards,
Jerry



From anthony.minessale at gmail.com  Tue Sep  4 21:15:13 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Sep 2012 12:15:13 -0500
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
Message-ID: <CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>

you are missing some of the sip and make sure its GIT HEAD (this
should be on jira)

sofia global siptrace on



On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
> Hi MC,
>
> Thanks a bunch for your reply, sorry for the delay.
> Here are my logs:
> http://pastebin.freeswitch.org/19831
>
> Any idea would be greatly appreciated.
>
> Best,
> Emrah
>
>
> On Aug 28, 2012, at 10:56 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>> 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Sep  4 21:17:30 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Sep 2012 12:17:30 -0500
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
Message-ID: <CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>

Oh,

And you should try to avoid doing an attended transfer to
one-legged-calls like conference or ivr, blind transfers work better
for this because calls to apps are not bridged and the concept of
transferring becomes confusing.  The other alternative is to bridge to
the conference by looping the call over loopback or calling to the
same box on sip so there is a true bridge.  But blind transferring is
the best solution.


On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you are missing some of the sip and make sure its GIT HEAD (this
> should be on jira)
>
> sofia global siptrace on
>
>
>
> On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
>> Hi MC,
>>
>> Thanks a bunch for your reply, sorry for the delay.
>> Here are my logs:
>> http://pastebin.freeswitch.org/19831
>>
>> Any idea would be greatly appreciated.
>>
>> Best,
>> Emrah
>>
>>
>> On Aug 28, 2012, at 10:56 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900



-- 
Anthony Minessale II

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

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

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


From mike.burlingame at me.com  Tue Sep  4 21:43:09 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Tue, 04 Sep 2012 10:43:09 -0700
Subject: [Freeswitch-users] Video Codec Sent In SDP Offer
In-Reply-To: <1545146083A72C4DB7B66584B7E5D9841D15BD5C@BY2PRD0410MB377.namprd04.prod.outlook.com>
References: <1545146083A72C4DB7B66584B7E5D9841D15BD5C@BY2PRD0410MB377.namprd04.prod.outlook.com>
Message-ID: <89005DB6-F21E-4B7B-B91E-8E8CE3D495B5@me.com>

Have you tried <param name="inbound-codec-negotiation" value="generous"/>

>From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#inbound-codec-negotiation

inbound-codec-negotiation
set to 'greedy' if you want your codec list to take precedence
<param name="inbound-codec-negotiation" value="generous"/>
if 'greedy' doesn't work for you, try 'scrooge' which has been known to fix misreported ptime issues with DID providers such as CallCentric.
A rule of thumb is:
'generous' permits the remote codec list have precedence and 'win' the codec negotiation and selection process
'greedy' forces a win by the local FreeSWITCH preference list
'scrooge' takes 'greedy' a step further, so that the FreeSWITCH wins even when the far side lies about capabilities during the negotiation process
sip_codec_negotiation is a channel variable version of this setting
inbound-late-negotiation
Uncomment to let calls hit the dialplan *before* you decide if the codec is OK.
<param name="inbound-late-negotiation" value="true"/>

Sent from my iPad

On Sep 4, 2012, at 9:53 AM, Jerry Richards <jerry.richards at teotech.com> wrote:

> If a call is made from a phone that does not include a video codec (i.e. H.264), Freeswitch still offers H.264 as a codec in the INVITE to the callee.  I think this is because H.264 is included in the inbound-codec-prefs and outbound-codec-prefs of the sip_profile (which is needed to support video calls).
> 
> Is there an easy way to configure Freeswitch to exclude H.264 based on the caller's INVITE?  Because it never would make sense for it to offer H.264 in this scenario.
> 
> Best Regards,
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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-users at vocalspace.com  Tue Sep  4 22:00:17 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Tue, 4 Sep 2012 13:00:17 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <CAKbxfG8vaTvAKfTUNMV+u=jm=6isqu2AVT=DHbVWcBzkbCfmpQ@mail.gmail.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
	<CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
	<16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
	<CAKbxfG8vaTvAKfTUNMV+u=jm=6isqu2AVT=DHbVWcBzkbCfmpQ@mail.gmail.com>
Message-ID: <87A42A1F-F4E3-40B0-88C7-5EFEE0AD328C@vocalspace.com>

I have tried to get the Current HEAD to run mod_managed with this to the top of switch_core.h
#define SWITCH_DEBUG_RWLOCKS 1

I am getting this error now. 

2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_managed.so
**/usr/local/freeswitch/mod/mod_managed.so: undefined symbol: switch_core_session_read_lock**

Thoughts?

Thanks!
On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:

> add this to the top of switch_core.h
> #define SWITCH_DEBUG_RWLOCKS 1
> 
> rebuild and get a full log of the call
> look for sign of unhandled rwlock
> 
> and put this on jira why I am i helping you over ml .... >=0
> 
> 
> On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
> <freeswitch-users at vocalspace.com> wrote:
>> Sorry Yes using the latest.
>> 
>> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date:   Thu Aug 30 17:17:15 2012 -0500
>> 
>> Changes made switch_cpp.cpp starting at Line 1000
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling CoreSession::destroy\n");
>>        if (session) {
>>        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have valid session\n");
>> 
>>                if (!channel) {
>>                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");
>> 
>>                        channel = switch_core_session_get_channel(session);
>>                }
>> 
>>                if (channel) {
>>                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>                                                          "%s destroy/unlink session from object\n", switch_channel_get_name(channel));
>>                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>                                                          "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>>                        switch_channel_set_private(channel, "CoreSession", NULL);
>>                        if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel, CF_TRANSFER)) {
>>                                switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>>                        }
>>                } else {
>>                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
>>                }
>> 
>> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>> [CRIT] switch_cpp.cpp:1002 We still have valid session
>> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink session from object
>> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9 destroy/unlink uuid
>> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
>> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX ) State REPORTING going to sleep
>> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX ) State Change CS_REPORTING -> CS_DESTROY
>> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX  [BREAK]
>> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX ) Locked, Waiting on external entities
>> 
>> 
>> 
>> channel remains hung
>> 
>> P
>> 
>> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>> 
>>> 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
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 nbhatti at gmail.com  Tue Sep  4 22:23:05 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 4 Sep 2012 21:23:05 +0300
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
Message-ID: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>

Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
NORMAL_CLEARING while the calls still have a valid duration and answered
time. For a typical billing scenario, calls are billed if (answered time)
billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the enumeration
would be other than normal and while the calls would be still a valid
answered call?

Thanks
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From nbhatti at gmail.com  Tue Sep  4 22:28:28 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 4 Sep 2012 21:28:28 +0300
Subject: [Freeswitch-users] Limitation
In-Reply-To: <CAKV-jFwcDit0hBz6Fhv9V9JWY78HxKERveWx1ZZXivbpNG_p3Q@mail.gmail.com>
References: <502B8B67.3090306@pripojtese.net>
	<CAMwZ-tJdeDXKt3qEBZaqSSgkfZBqKS0+bDwugewqLGrkyc_5mg@mail.gmail.com>
	<CAJUJwtgda662nZQgx6p7Nyjq_JmzHuCwMnuxRWWv5OMe=8VnFA@mail.gmail.com>
	<CAKV-jFwcDit0hBz6Fhv9V9JWY78HxKERveWx1ZZXivbpNG_p3Q@mail.gmail.com>
Message-ID: <CACak_yzz7HEA3FOeECzcFKt3ycJqKK-5y2UNxBQ0uyha5ds0eA@mail.gmail.com>

These are call limitations in current version, however, this is not
something which is going to be there forever. We are working on v2 which is
going to be released under MPL v2 with full source around end of this year.
Current version does not have the LUA source available. btw: These limits
have nothing to do with FreeSWITCH not they advertise or limit any
capability of FreeSWITCH.

Muhammad

On Tue, Sep 4, 2012 at 1:04 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> It is a limitation imposed by vbilling. However it isn't one you are
> supposed to be able to change (easily). The author has decided to impose
> this limitation for his own reasons, and my intent is to not speak for him
> here.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Sep 4, 2012 5:29 AM, "SamyGo" <govoiper at gmail.com> wrote:
>
>> Hi,
>>
>> " Usage for CONCURRENT_CALLS_IN_FREE_MODE_calls_max is now 0 " - Thats
>> definitely not FreeSwitch. just find this particular variable in vbilling
>> or fusionpbx code and try increasing the decrementing counter.
>>
>> Thanks,
>> Sammy
>>
>>
>>
>> On Mon, Sep 3, 2012 at 12:59 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>> On Wed, Aug 15, 2012 at 5:43 AM, Jakub Tencl <info at pripojtese.net>
>>> wrote:
>>> > 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?
>>>
>>> Wouldn't the limitation be on the vBilling side (if there is any)? It
>>> doesn't seem like this list is the right place to ask. Anyway, good
>>> luck and let us know what you learn.
>>>
>>>
>>> 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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nbhatti at gmail.com  Tue Sep  4 22:31:08 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 4 Sep 2012 21:31:08 +0300
Subject: [Freeswitch-users] Limitation
In-Reply-To: <CAKV-jFwcDit0hBz6Fhv9V9JWY78HxKERveWx1ZZXivbpNG_p3Q@mail.gmail.com>
References: <502B8B67.3090306@pripojtese.net>
	<CAMwZ-tJdeDXKt3qEBZaqSSgkfZBqKS0+bDwugewqLGrkyc_5mg@mail.gmail.com>
	<CAJUJwtgda662nZQgx6p7Nyjq_JmzHuCwMnuxRWWv5OMe=8VnFA@mail.gmail.com>
	<CAKV-jFwcDit0hBz6Fhv9V9JWY78HxKERveWx1ZZXivbpNG_p3Q@mail.gmail.com>
Message-ID: <CACak_yzQ4MukHSy7Hky18Ab9j29E_1Y7qO41bPOiwF5rX+6rxA@mail.gmail.com>

.. and yes, the limitation might be changeable (easily) but won't this be
breaking the license agreement? hint hint .. :)

On Tue, Sep 4, 2012 at 1:04 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> It is a limitation imposed by vbilling. However it isn't one you are
> supposed to be able to change (easily). The author has decided to impose
> this limitation for his own reasons, and my intent is to not speak for him
> here.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Sep 4, 2012 5:29 AM, "SamyGo" <govoiper at gmail.com> wrote:
>
>> Hi,
>>
>> " Usage for CONCURRENT_CALLS_IN_FREE_MODE_calls_max is now 0 " - Thats
>> definitely not FreeSwitch. just find this particular variable in vbilling
>> or fusionpbx code and try increasing the decrementing counter.
>>
>> Thanks,
>> Sammy
>>
>>
>>
>> On Mon, Sep 3, 2012 at 12:59 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>> On Wed, Aug 15, 2012 at 5:43 AM, Jakub Tencl <info at pripojtese.net>
>>> wrote:
>>> > 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?
>>>
>>> Wouldn't the limitation be on the vBilling side (if there is any)? It
>>> doesn't seem like this list is the right place to ask. Anyway, good
>>> luck and let us know what you learn.
>>>
>>>
>>> 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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Sep  4 22:37:54 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Sep 2012 13:37:54 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <87A42A1F-F4E3-40B0-88C7-5EFEE0AD328C@vocalspace.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
	<CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
	<16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
	<CAKbxfG8vaTvAKfTUNMV+u=jm=6isqu2AVT=DHbVWcBzkbCfmpQ@mail.gmail.com>
	<87A42A1F-F4E3-40B0-88C7-5EFEE0AD328C@vocalspace.com>
Message-ID: <CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>

now that you have a jira do not continue this thread it doubles the
work effort, see comments there

On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
<freeswitch-users at vocalspace.com> wrote:
> I have tried to get the Current HEAD to run mod_managed with this to the top of switch_core.h
> #define SWITCH_DEBUG_RWLOCKS 1
>
> I am getting this error now.
>
> 2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_managed.so
> **/usr/local/freeswitch/mod/mod_managed.so: undefined symbol: switch_core_session_read_lock**
>
> Thoughts?
>
> Thanks!
> On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:
>
>> add this to the top of switch_core.h
>> #define SWITCH_DEBUG_RWLOCKS 1
>>
>> rebuild and get a full log of the call
>> look for sign of unhandled rwlock
>>
>> and put this on jira why I am i helping you over ml .... >=0
>>
>>
>> On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
>> <freeswitch-users at vocalspace.com> wrote:
>>> Sorry Yes using the latest.
>>>
>>> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> Date:   Thu Aug 30 17:17:15 2012 -0500
>>>
>>> Changes made switch_cpp.cpp starting at Line 1000
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling CoreSession::destroy\n");
>>>        if (session) {
>>>        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have valid session\n");
>>>
>>>                if (!channel) {
>>>                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");
>>>
>>>                        channel = switch_core_session_get_channel(session);
>>>                }
>>>
>>>                if (channel) {
>>>                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>                                                          "%s destroy/unlink session from object\n", switch_channel_get_name(channel));
>>>                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>                                                          "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>>>                        switch_channel_set_private(channel, "CoreSession", NULL);
>>>                        if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel, CF_TRANSFER)) {
>>>                                switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>>>                        }
>>>                } else {
>>>                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
>>>                }
>>>
>>> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>>> [CRIT] switch_cpp.cpp:1002 We still have valid session
>>> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink session from object
>>> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9 destroy/unlink uuid
>>> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
>>> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX ) State REPORTING going to sleep
>>> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX ) State Change CS_REPORTING -> CS_DESTROY
>>> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX  [BREAK]
>>> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX ) Locked, Waiting on external entities
>>>
>>>
>>>
>>> channel remains hung
>>>
>>> P
>>>
>>> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>>>
>>>> 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
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 lists at kavun.ch  Tue Sep  4 22:52:27 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 4 Sep 2012 14:52:27 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
Message-ID: <3BFA0C81-168F-441D-9D55-D69DB58D8436@kavun.ch>

Anthony,

Thank you for taking the time to provide your input. I did not think the referrer's side of SIP was relevant, and so enabled SIPtrace on the external peer only.
I will be doing more tests and keep you posted.

It feels as if I underwent a complete telephony mindset upgrade ever since I traded Asterisk for FS. Thank you for making it happen.

All the best,
Emrah

On Sep 4, 2012, at 1:15 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> you are missing some of the sip and make sure its GIT HEAD (this
> should be on jira)
> 
> sofia global siptrace on
> 
> 
> 
> On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
>> Hi MC,
>> 
>> Thanks a bunch for your reply, sorry for the delay.
>> Here are my logs:
>> http://pastebin.freeswitch.org/19831
>> 
>> Any idea would be greatly appreciated.
>> 
>> Best,
>> Emrah
>> 
>> 
>> On Aug 28, 2012, at 10:56 AM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> 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
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 freeswitch-users at vocalspace.com  Tue Sep  4 23:05:54 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Tue, 4 Sep 2012 14:05:54 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
	<CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
	<16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
	<CAKbxfG8vaTvAKfTUNMV+u=jm=6isqu2AVT=DHbVWcBzkbCfmpQ@mail.gmail.com>
	<87A42A1F-F4E3-40B0-88C7-5EFEE0AD328C@vocalspace.com>
	<CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>
Message-ID: <FE33CF4A-C315-4A04-8545-3FAB25AF863D@vocalspace.com>

Sorry for the delay in the JIRA. I missed the last email before going on holiday.

Regards,

Phillip
On Sep 4, 2012, at 1:37 PM, Anthony Minessale wrote:

> now that you have a jira do not continue this thread it doubles the
> work effort, see comments there
> 
> On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
> <freeswitch-users at vocalspace.com> wrote:
>> I have tried to get the Current HEAD to run mod_managed with this to the top of switch_core.h
>> #define SWITCH_DEBUG_RWLOCKS 1
>> 
>> I am getting this error now.
>> 
>> 2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_managed.so
>> **/usr/local/freeswitch/mod/mod_managed.so: undefined symbol: switch_core_session_read_lock**
>> 
>> Thoughts?
>> 
>> Thanks!
>> On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:
>> 
>>> add this to the top of switch_core.h
>>> #define SWITCH_DEBUG_RWLOCKS 1
>>> 
>>> rebuild and get a full log of the call
>>> look for sign of unhandled rwlock
>>> 
>>> and put this on jira why I am i helping you over ml .... >=0
>>> 
>>> 
>>> On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
>>> <freeswitch-users at vocalspace.com> wrote:
>>>> Sorry Yes using the latest.
>>>> 
>>>> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
>>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>> Date:   Thu Aug 30 17:17:15 2012 -0500
>>>> 
>>>> Changes made switch_cpp.cpp starting at Line 1000
>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling CoreSession::destroy\n");
>>>>       if (session) {
>>>>       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have valid session\n");
>>>> 
>>>>               if (!channel) {
>>>>                       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");
>>>> 
>>>>                       channel = switch_core_session_get_channel(session);
>>>>               }
>>>> 
>>>>               if (channel) {
>>>>                       switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>>                                                         "%s destroy/unlink session from object\n", switch_channel_get_name(channel));
>>>>                       switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>>                                                         "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>>>>                       switch_channel_set_private(channel, "CoreSession", NULL);
>>>>                       if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel, CF_TRANSFER)) {
>>>>                               switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>>>>                       }
>>>>               } else {
>>>>                       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
>>>>               }
>>>> 
>>>> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>>>> [CRIT] switch_cpp.cpp:1002 We still have valid session
>>>> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink session from object
>>>> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9 destroy/unlink uuid
>>>> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
>>>> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX ) State REPORTING going to sleep
>>>> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX ) State Change CS_REPORTING -> CS_DESTROY
>>>> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX  [BREAK]
>>>> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX ) Locked, Waiting on external entities
>>>> 
>>>> 
>>>> 
>>>> channel remains hung
>>>> 
>>>> P
>>>> 
>>>> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>>>> 
>>>>> 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
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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



From drk at drkngs.net  Tue Sep  4 23:08:58 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 04 Sep 2012 12:08:58 -0700
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>
Message-ID: <20120904190858.cc516138@mail.tritonwest.net>

This may be a bug, but ManagedSession really is not the right way to do this from API context. Because of the APP_DOMAIN issue its much easier to just do Api.ExecuteString("originate ... "). If you need to get to the "managed session of the leg you have two options:  
   
For getting to it before the originate set the variable execute_on_originate to call a managed AppPlugin, or:  
   
For getting results at the end of the call in API code, set the variable API_HANGUP_HOOK.   
   
Both of these methoods are much easier to do from API context in mod_managed, and you won't have to worry about crossing app domain boundries, and you won't have do do any cleanup on the leg.
      _____  

  From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 04 Sep 2012 11:37:54 -0700
Subject: Re: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state

now that you have a jira do not continue this thread it doubles the
work effort, see comments there

On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
<freeswitch-users at vocalspace.com> wrote:
> I have tried to get the Current HEAD to run mod_managed with this to the top of switch_core.h
> #define SWITCH_DEBUG_RWLOCKS 1
>
> I am getting this error now.
>
> 2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_managed.so
> **/usr/local/freeswitch/mod/mod_managed.so: undefined symbol: switch_core_session_read_lock**
>
> Thoughts?
>
> Thanks!
> On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:
>
>> add this to the top of switch_core.h
>> #define SWITCH_DEBUG_RWLOCKS 1
>>
>> rebuild and get a full log of the call
>> look for sign of unhandled rwlock
>>
>> and put this on jira why I am i helping you over ml .... >=0
>>
>>
>> On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
>> <freeswitch-users at vocalspace.com> wrote:
>>> Sorry Yes using the latest.
>>>
>>> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> Date: Thu Aug 30 17:17:15 2012 -0500
>>>
>>> Changes made switch_cpp.cpp starting at Line 1000
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling CoreSession::destroy\n");
>>> if (session) {
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have valid session\n");
>>>
>>> if (!channel) {
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! Trying to get it!\n");
>>>
>>> channel = switch_core_session_get_channel(session);
>>> }
>>>
>>> if (channel) {
>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>> "%s destroy/unlink session from object\n", switch_channel_get_name(channel));
>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>> "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>>> switch_channel_set_private(channel, "CoreSession", NULL);
>>> if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) && !switch_channel_test_flag(channel, CF_TRANSFER)) {
>>> switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>>> }
>>> } else {
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is undefined! We Failed to get it!\n");
>>> }
>>>
>>> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>>> [CRIT] switch_cpp.cpp:1002 We still have valid session
>>> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink session from object
>>> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9 destroy/unlink uuid
>>> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
>>> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX ) State REPORTING going to sleep
>>> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX ) State Change CS_REPORTING -> CS_DESTROY
>>> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX [BREAK]
>>> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX ) Locked, Waiting on external entities
>>>
>>>
>>>
>>> channel remains hung
>>>
>>> P
>>>
>>> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>>>
>>>> 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
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 sdevoy at bizfocused.com  Tue Sep  4 23:54:56 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 4 Sep 2012 15:54:56 -0400
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
Message-ID: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>

Hi,

 

One of my customers has asked if we can setup a phone on an ATA (or
otherwise if necessary) that you just pick up and it is automatically
connected to some number?  Much like the "Bat-phone" or the US to USSR
Emergency Phone in the Situation room out the White House.

 

Thoughts?

 

Thanks,

Sean

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From marketing at cluecon.com  Wed Sep  5 00:00:51 2012
From: marketing at cluecon.com (Michael Collins)
Date: Tue, 4 Sep 2012 13:00:51 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxWxtFRbL1idYh=XBHNErx3obT8nKwamCstFsSSSHv9nwg@mail.gmail.com>

Happy Tuesday to you all! I hope you had a relaxing 3-day weekend.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_29>
 we had a nice discussion about TLS <http://wiki.freeswitch.org/wiki/Tls> in
FreeSWITCH. Mitch Capper helped demystify some of the things that go on
with creating certificates, handling certificate authorities (CA), and the
like. This week we will be having a community discussion on how to improve
our handling of poorly-worded or inappropriate questions sent to the
mailing list. Ourupcoming
schedule<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_29#Upcoming_Presentations>
 for September is all booked up, so make plans to be on our Wednesday
conference calls.

Be on the lookout for updated English Callie and French June sounds. They
have been recorded and are being verified. I suspect they will show up in
the next day or two.

The new FreeSWITCH book is moving along as well. I just submitted chapter 2
to our publisher and am now on to redrafting chapter 3. On a side note I
just want to mention that I updated the FreeSWITCH Wikipedia
page<http://en.wikipedia.org/wiki/FreeSWITCH>
 to reflect the fact that we are now on version 1.2.1 and that we've formed
stable and development branches.

ClueCon 2012 attendees and FreeSWITCH community members qualify for special
pricing at the IIT Real-Time Communications Conference and
Expo<http://www.rtc-conference.com/register/>.
Use the special discount code *FREWDSC* when you register. More details are
available at ClueCon.com <http://cluecon.com/>.

Have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From msc at freeswitch.org  Wed Sep  5 00:04:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Sep 2012 13:04:30 -0700
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <3BFA0C81-168F-441D-9D55-D69DB58D8436@kavun.ch>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<3BFA0C81-168F-441D-9D55-D69DB58D8436@kavun.ch>
Message-ID: <CAKzWOxVePFHNGHjWVKcPgKheHbtMu=715s5vaaoNN8B8Z63jNw@mail.gmail.com>

> It feels as if I underwent a complete telephony mindset upgrade ever since
> I traded Asterisk for FS. Thank you for making it happen.
>
> This process is known as pulling your head out of your Asterisk...
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From grcamauer at gmail.com  Wed Sep  5 00:10:17 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 4 Sep 2012 17:10:17 -0300
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <1346730034461-7582520.post@n2.nabble.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
Message-ID: <CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>

Thank you mazilo and Jeff for the fix.  By looking at the GIT log I see
that spandsp and libtiff are undergoing major changes these days.  I will
stick to STABLE for a couple of weeks while I monitor progress on current.



On Tue, Sep 4, 2012 at 12:40 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> I re-added the missing configure.gnu file.
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Missing-libtiff-after-make-current-tp7582512p7582520.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
>



-- 
Guillermo Ruiz Camauer
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From anthony.minessale at gmail.com  Wed Sep  5 00:10:22 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Sep 2012 15:10:22 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <20120904190858.cc516138@mail.tritonwest.net>
References: <CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>
	<20120904190858.cc516138@mail.tritonwest.net>
Message-ID: <CAKbxfG9UL4zh1w5xTLoxqBzEHBJ94MRbY+6hvJu21Ze21ZCqsQ@mail.gmail.com>

Listen to Dave.  I fixed the lock leak, it was down deep into code I
am not sure is ever used.   You want to be careful with what you do
especially since you are on linux which means you must be using Mono
which I am also not sure is used a lot.



On Tue, Sep 4, 2012 at 2:08 PM, Dave R. Kompel <drk at drkngs.net> wrote:
> This may be a bug, but ManagedSession really is not the right way to do this
> from API context. Because of the APP_DOMAIN issue its much easier to just do
> Api.ExecuteString("originate ... "). If you need to get to the "managed
> session of the leg you have two options:
>
> For getting to it before the originate set the variable execute_on_originate
> to call a managed AppPlugin, or:
>
> For getting results at the end of the call in API code, set the variable
> API_HANGUP_HOOK.
>
> Both of these methoods are much easier to do from API context in
> mod_managed, and you won't have to worry about crossing app domain
> boundries, and you won't have do do any cleanup on the leg.
>
> ________________________________
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Tue, 04 Sep 2012 11:37:54 -0700
>
> Subject: Re: [Freeswitch-users] Problem with originated calls hanging in
> CS_SOFT_EXEC state
>
> now that you have a jira do not continue this thread it doubles the
> work effort, see comments there
>
> On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
> <freeswitch-users at vocalspace.com> wrote:
>> I have tried to get the Current HEAD to run mod_managed with this to the
>> top of switch_core.h
>> #define SWITCH_DEBUG_RWLOCKS 1
>>
>> I am getting this error now.
>>
>> 2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error
>> Loading module /usr/local/freeswitch/mod/mod_managed.so
>> **/usr/local/freeswitch/mod/mod_managed.so: undefined symbol:
>> switch_core_session_read_lock**
>>
>> Thoughts?
>>
>> Thanks!
>> On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:
>>
>>> add this to the top of switch_core.h
>>> #define SWITCH_DEBUG_RWLOCKS 1
>>>
>>> rebuild and get a full log of the call
>>> look for sign of unhandled rwlock
>>>
>>> and put this on jira why I am i helping you over ml .... >=0
>>>
>>>
>>> On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
>>> <freeswitch-users at vocalspace.com> wrote:
>>>> Sorry Yes using the latest.
>>>>
>>>> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
>>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>> Date: Thu Aug 30 17:17:15 2012 -0500
>>>>
>>>> Changes made switch_cpp.cpp starting at Line 1000
>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling
>>>> CoreSession::destroy\n");
>>>> if (session) {
>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have
>>>> valid session\n");
>>>>
>>>> if (!channel) {
>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
>>>> undefined! Trying to get it!\n");
>>>>
>>>> channel = switch_core_session_get_channel(session);
>>>> }
>>>>
>>>> if (channel) {
>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>> "%s destroy/unlink session from object\n",
>>>> switch_channel_get_name(channel));
>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>> "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>>>> switch_channel_set_private(channel, "CoreSession", NULL);
>>>> if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) &&
>>>> !switch_channel_test_flag(channel, CF_TRANSFER)) {
>>>> switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>>>> }
>>>> } else {
>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
>>>> undefined! We Failed to get it!\n");
>>>> }
>>>>
>>>> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>>>> [CRIT] switch_cpp.cpp:1002 We still have valid session
>>>> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink
>>>> session from object
>>>> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9
>>>> destroy/unlink uuid
>>>> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX
>>>> Standard REPORTING, cause: NORMAL_CLEARING
>>>> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX )
>>>> State REPORTING going to sleep
>>>> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX )
>>>> State Change CS_REPORTING -> CS_DESTROY
>>>> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX
>>>> [BREAK]
>>>> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX
>>>> ) Locked, Waiting on external entities
>>>>
>>>>
>>>>
>>>> channel remains hung
>>>>
>>>> P
>>>>
>>>> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>>>>
>>>>> 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
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From avi at avimarcus.net  Wed Sep  5 00:11:23 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 4 Sep 2012 23:11:23 +0300
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
In-Reply-To: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
References: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
Message-ID: <CALNZuw=cCdgRAnWs97BwoHWZd3+HxCYRhLy7y-_eC-Mg8keyAg@mail.gmail.com>

SPA-2102's do it and I presume anything else with it's dialing plan.

See:
http://www.cisco.com/en/US/products/ps10030/products_qanda_item09186a0080a36559.shtml
question
#5.

-Avi


On Tue, Sep 4, 2012 at 10:54 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,****
>
> ** **
>
> One of my customers has asked if we can setup a phone on an ATA (or
> otherwise if necessary) that you just pick up and it is automatically
> connected to some number?  Much like the ?Bat-phone? or the US to USSR
> Emergency Phone in the Situation room out the White House.****
>
> ** **
>
> Thoughts?****
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep  5 00:17:26 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 4 Sep 2012 14:17:26 -0600
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
In-Reply-To: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
References: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
Message-ID: <CAMwZ-t+66=n992kX4sXwVYE6dx25iCqgLTwBLx27xgz+LMoWkw@mail.gmail.com>

On Tue, Sep 4, 2012 at 1:54 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> One of my customers has asked if we can setup a phone on an ATA (or
> otherwise if necessary) that you just pick up and it is automatically
> connected to some number?  Much like the ?Bat-phone? or the US to USSR
> Emergency Phone in the Situation room out the White House.

Without looking into this, I'd say you're looking at a feature on the
SIP phone. The servers don't know if the line was picked up or not.
So, you need hardware or software that says, "Now that the phone was
taken off the hook, call this number." Once you have that, you can
solve the rest of the problem in your dialplan.


Gabe


From grcamauer at gmail.com  Wed Sep  5 00:20:35 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 4 Sep 2012 17:20:35 -0300
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
In-Reply-To: <CALNZuw=cCdgRAnWs97BwoHWZd3+HxCYRhLy7y-_eC-Mg8keyAg@mail.gmail.com>
References: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
	<CALNZuw=cCdgRAnWs97BwoHWZd3+HxCYRhLy7y-_eC-Mg8keyAg@mail.gmail.com>
Message-ID: <CAPNTnnogzNSi2Qox+-y1bfnhwhzRKrO2o9=O7zORHrB5a3gg9g@mail.gmail.com>

Grandstream HT-502 ATA has: *Offhook Auto-Dial: *  (User ID/extension to
dial automatically when offhook)

Which will do what you want.

On Tue, Sep 4, 2012 at 5:11 PM, Avi Marcus <avi at avimarcus.net> wrote:

> SPA-2102's do it and I presume anything else with it's dialing plan.
>
> See:
> http://www.cisco.com/en/US/products/ps10030/products_qanda_item09186a0080a36559.shtml question
> #5.
>
> -Avi
>
>
> On Tue, Sep 4, 2012 at 10:54 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> Hi,****
>>
>> ** **
>>
>> One of my customers has asked if we can setup a phone on an ATA (or
>> otherwise if necessary) that you just pick up and it is automatically
>> connected to some number?  Much like the ?Bat-phone? or the US to USSR
>> Emergency Phone in the Situation room out the White House.****
>>
>> ** **
>>
>> Thoughts?****
>>
>> ** **
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch-users at vocalspace.com  Wed Sep  5 00:35:47 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Tue, 4 Sep 2012 15:35:47 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <CAKbxfG9UL4zh1w5xTLoxqBzEHBJ94MRbY+6hvJu21Ze21ZCqsQ@mail.gmail.com>
References: <CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>
	<20120904190858.cc516138@mail.tritonwest.net>
	<CAKbxfG9UL4zh1w5xTLoxqBzEHBJ94MRbY+6hvJu21Ze21ZCqsQ@mail.gmail.com>
Message-ID: <6D0900A6-2881-47EF-8FEB-CC7354EA786E@vocalspace.com>

Thanks Anthony! I found the ManageSession methods by looking at the code that is exposed via swig.

Using  "execute_on_*" hooks seems to be the proper procedure, it needs to be documented on the wiki. There is virtually no documentation to originate a call for mod managed on the wiki. I will correct that if I can get wiki access.

I will illustrate your solution using "execute_on_originate" as the preferred solution. There are also several "execute_on_*" hooks that would work. 

Should ManagedSession.OriginateHandleHangup and .Originate() methods be deprecated to discourage their use in further releases? Or at least some comments in the code that point people to look at execute_on directives. Does Java have this same issue?

Anthony,Glad I found a bug by doing the wrong thing.... sorry for making work for ya!

I cannot say enough about how responsive Anthony and the community has been about this issue. THANKS!

On Sep 4, 2012, at 3:10 PM, Anthony Minessale wrote:

> Listen to Dave.  I fixed the lock leak, it was down deep into code I
> am not sure is ever used.   You want to be careful with what you do
> especially since you are on linux which means you must be using Mono
> which I am also not sure is used a lot.
> 
> 
> 
> On Tue, Sep 4, 2012 at 2:08 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>> This may be a bug, but ManagedSession really is not the right way to do this
>> from API context. Because of the APP_DOMAIN issue its much easier to just do
>> Api.ExecuteString("originate ... "). If you need to get to the "managed
>> session of the leg you have two options:
>> 
>> For getting to it before the originate set the variable execute_on_originate
>> to call a managed AppPlugin, or:
>> 
>> For getting results at the end of the call in API code, set the variable
>> API_HANGUP_HOOK.
>> 
>> Both of these methoods are much easier to do from API context in
>> mod_managed, and you won't have to worry about crossing app domain
>> boundries, and you won't have do do any cleanup on the leg.
>> 
>> ________________________________
>> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
>> Sent: Tue, 04 Sep 2012 11:37:54 -0700
>> 
>> Subject: Re: [Freeswitch-users] Problem with originated calls hanging in
>> CS_SOFT_EXEC state
>> 
>> now that you have a jira do not continue this thread it doubles the
>> work effort, see comments there
>> 
>> On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
>> <freeswitch-users at vocalspace.com> wrote:
>>> I have tried to get the Current HEAD to run mod_managed with this to the
>>> top of switch_core.h
>>> #define SWITCH_DEBUG_RWLOCKS 1
>>> 
>>> I am getting this error now.
>>> 
>>> 2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error
>>> Loading module /usr/local/freeswitch/mod/mod_managed.so
>>> **/usr/local/freeswitch/mod/mod_managed.so: undefined symbol:
>>> switch_core_session_read_lock**
>>> 
>>> Thoughts?
>>> 
>>> Thanks!
>>> On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:
>>> 
>>>> add this to the top of switch_core.h
>>>> #define SWITCH_DEBUG_RWLOCKS 1
>>>> 
>>>> rebuild and get a full log of the call
>>>> look for sign of unhandled rwlock
>>>> 
>>>> and put this on jira why I am i helping you over ml .... >=0
>>>> 
>>>> 
>>>> On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
>>>> <freeswitch-users at vocalspace.com> wrote:
>>>>> Sorry Yes using the latest.
>>>>> 
>>>>> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
>>>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>>> Date: Thu Aug 30 17:17:15 2012 -0500
>>>>> 
>>>>> Changes made switch_cpp.cpp starting at Line 1000
>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling
>>>>> CoreSession::destroy\n");
>>>>> if (session) {
>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have
>>>>> valid session\n");
>>>>> 
>>>>> if (!channel) {
>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
>>>>> undefined! Trying to get it!\n");
>>>>> 
>>>>> channel = switch_core_session_get_channel(session);
>>>>> }
>>>>> 
>>>>> if (channel) {
>>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>>> "%s destroy/unlink session from object\n",
>>>>> switch_channel_get_name(channel));
>>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>>> "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>>>>> switch_channel_set_private(channel, "CoreSession", NULL);
>>>>> if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) &&
>>>>> !switch_channel_test_flag(channel, CF_TRANSFER)) {
>>>>> switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>>>>> }
>>>>> } else {
>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
>>>>> undefined! We Failed to get it!\n");
>>>>> }
>>>>> 
>>>>> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>>>>> [CRIT] switch_cpp.cpp:1002 We still have valid session
>>>>> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink
>>>>> session from object
>>>>> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9
>>>>> destroy/unlink uuid
>>>>> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX
>>>>> Standard REPORTING, cause: NORMAL_CLEARING
>>>>> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX )
>>>>> State REPORTING going to sleep
>>>>> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX )
>>>>> State Change CS_REPORTING -> CS_DESTROY
>>>>> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX
>>>>> [BREAK]
>>>>> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX
>>>>> ) Locked, Waiting on external entities
>>>>> 
>>>>> 
>>>>> 
>>>>> channel remains hung
>>>>> 
>>>>> P
>>>>> 
>>>>> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 jerry.richards at teotech.com  Wed Sep  5 00:36:44 2012
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 4 Sep 2012 20:36:44 +0000
Subject: [Freeswitch-users] Video Codec Sent In SDP Offer
In-Reply-To: <89005DB6-F21E-4B7B-B91E-8E8CE3D495B5@me.com>
References: <1545146083A72C4DB7B66584B7E5D9841D15BD5C@BY2PRD0410MB377.namprd04.prod.outlook.com>
	<89005DB6-F21E-4B7B-B91E-8E8CE3D495B5@me.com>
Message-ID: <1545146083A72C4DB7B66584B7E5D9841D15DECC@BY2PRD0410MB377.namprd04.prod.outlook.com>

Actually, the issue is introduced prior to SDP negotiation.  That is, Freeswitch includes video codecs in the very first SDP offer sent to the callee when it shouldn't.

CONFIGURATION:
Phone A does not have video capability
Phone B does have video capablity
The conf/sip_profile/internal.xml has the following tags:
<param name="inbound-codec-negotiation" value="generous"/
<param name="inbound-bypass-media" value="false"/>
<param name="inbound-late-negotiation" value="true"/>

SCENARIO:
1. Phone A calls Phone B.
2. INVITE from Phone A to FS does not include video codecs.
3. INVITE from FS to Phone B includes video codecs (added by FS from the global codec list)
4. Phone B supports video codecs and presents the call as a video call, since that option is included in the codec list.
5. When video call is answered by Phone B, a video connection cannot be established because it is not supported by Phone A.

The problem is not related to who gets precedence in selecting the codec; the problem is that video codecs should not be offered by FS at all, since they cannot be supported by the caller.

Thanks,
Jerry


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Burlingame
Sent: Tuesday, September 04, 2012 10:43 AM
To: FreeSWITCH Users Help
Cc: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Video Codec Sent In SDP Offer

Have you tried <param name="inbound-codec-negotiation" value="generous"/>

>From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#inbound-codec-negotiation

inbound-codec-negotiation

set to 'greedy' if you want your codec list to take precedence

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

if 'greedy' doesn't work for you, try 'scrooge' which has been known to fix misreported ptime issues with DID providers such as CallCentric.

A rule of thumb is:
?  'generous' permits the remote codec list have precedence and 'win' the codec negotiation and selection process
?  'greedy' forces a win by the local FreeSWITCH preference list
?  'scrooge' takes 'greedy' a step further, so that the FreeSWITCH wins even when the far side lies about capabilities during the negotiation process

sip_codec_negotiation<http://wiki.freeswitch.org/wiki/Variable_sip_codec_negotiation> is a channel variable version of this setting

inbound-late-negotiation

Uncomment to let calls hit the dialplan *before* you decide if the codec is OK.

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

Sent from my iPad

On Sep 4, 2012, at 9:53 AM, Jerry Richards <jerry.richards at teotech.com<mailto:jerry.richards at teotech.com>> wrote:
If a call is made from a phone that does not include a video codec (i.e. H.264), Freeswitch still offers H.264 as a codec in the INVITE to the callee.  I think this is because H.264 is included in the inbound-codec-prefs and outbound-codec-prefs of the sip_profile (which is needed to support video calls).

Is there an easy way to configure Freeswitch to exclude H.264 based on the caller's INVITE?  Because it never would make sense for it to offer H.264 in this scenario.

Best Regards,
Jerry


_________________________________________________________________________
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
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From curriegrad2004 at gmail.com  Wed Sep  5 00:48:48 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 4 Sep 2012 13:48:48 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxWxtFRbL1idYh=XBHNErx3obT8nKwamCstFsSSSHv9nwg@mail.gmail.com>
References: <CAKzWOxWxtFRbL1idYh=XBHNErx3obT8nKwamCstFsSSSHv9nwg@mail.gmail.com>
Message-ID: <CAMWiMy8M6VeA-XqFxtP71a3hnBJFbpnE_sE_2xmopWnm7N=TjA@mail.gmail.com>

Need to add a discussion thing about people using master branch and
why you shouldn't do it for production. I've answered a few ML
questions regarding the broken spandsp code that caused master to fail
to build on all the platforms.

On Tue, Sep 4, 2012 at 1:00 PM, Michael Collins <marketing at cluecon.com> wrote:
> Happy Tuesday to you all! I hope you had a relaxing 3-day weekend.
>
> On last week's conference call we had a nice discussion about TLS in
> FreeSWITCH. Mitch Capper helped demystify some of the things that go on with
> creating certificates, handling certificate authorities (CA), and the like.
> This week we will be having a community discussion on how to improve our
> handling of poorly-worded or inappropriate questions sent to the mailing
> list. Ourupcoming schedule for September is all booked up, so make plans to
> be on our Wednesday conference calls.
>
> Be on the lookout for updated English Callie and French June sounds. They
> have been recorded and are being verified. I suspect they will show up in
> the next day or two.
>
> The new FreeSWITCH book is moving along as well. I just submitted chapter 2
> to our publisher and am now on to redrafting chapter 3. On a side note I
> just want to mention that I updated the FreeSWITCH Wikipedia page to reflect
> the fact that we are now on version 1.2.1 and that we've formed stable and
> development branches.
>
> ClueCon 2012 attendees and FreeSWITCH community members qualify for special
> pricing at the IIT Real-Time Communications Conference and Expo. Use the
> special discount code FREWDSC when you register. More details are available
> at ClueCon.com.
>
> Have a great week!
>
> --
> Michael S Collins
> ClueCon Team
> http://www.cluecon.com
> 877-7-4ACLUE
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed Sep  5 00:50:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Sep 2012 13:50:01 -0700
Subject: [Freeswitch-users] routing error????
In-Reply-To: <003801cd8ab3$9a255790$ce7006b0$@accra.ca>
References: <003801cd8ab3$9a255790$ce7006b0$@accra.ca>
Message-ID: <CAKzWOxVrZ=2uWfTkZwzu3Jyp+8GJak2Gikgfdj0x1eKS6Sc5dA@mail.gmail.com>

Get a siptrace along with the full debug log and drop it into pastebin.
Hopefully the siptrace will yield some additional clues...
-MC

On Tue, Sep 4, 2012 at 8:40 AM, Charles Bujold <cjbujold at accra.ca> wrote:

> This morning I am encountering the following error:****
>
> ** **
>
> The originating call is from extension 250  at 192.168.20.23  to extension
> 425 at 192.168.25.13 ( same server two different office locations).  from
> what I see I think it is some routing issue but not certain.  How can I
> determine what is the cause? And how can I fix it?****
>
> ** **
>
> Thanks****
>
> cjb****
>
> ** **
>
> ** **
>
> 2012-09-04 12:17:44.001046 [DEBUG] switch_core_session.c:1229 Send signal
> sofia/internal/sip:425 at 192.168.25.13:5062 [BREAK]****
>
> 2012-09-04 12:17:44.001046 [DEBUG] switch_core_state_machine.c:385
> (sofia/internal/sip:425 at 192.168.25.13:5062) Running State Change CS_HANGUP
> ****
>
> 2012-09-04 12:17:44.001046 [NOTICE] switch_ivr_originate.c:2544 Cannot
> create outgoing channel of type [user] cause: [NO_ANSWER]****
>
> 2012-09-04 12:17:44.001046 [DEBUG] switch_ivr_originate.c:3458 Originate
> Resulted in Error Cause: 19 [NO_ANSWER]****
>
> 2012-09-04 12:17:44.001046 [INFO] mod_dptools.c:3027 Originate Failed.
> Cause: NO_ANSWER****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep  5 00:50:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Sep 2012 13:50:58 -0700
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
In-Reply-To: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
References: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
Message-ID: <CAKzWOxUwi1wEYcKPhotQT95S_pdGVpL2WEdHYk+AVbv8pS1nSg@mail.gmail.com>

Do you have some call examples we could review?
-MC

On Tue, Sep 4, 2012 at 11:23 AM, Muhammad Naseer Bhatti
<nbhatti at gmail.com>wrote:

> Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
> NORMAL_CLEARING while the calls still have a valid duration and answered
> time. For a typical billing scenario, calls are billed if (answered time)
> billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the enumeration
> would be other than normal and while the calls would be still a valid
> answered call?
>
> 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
>
>


-- 
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 Sep  5 01:02:29 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Sep 2012 14:02:29 -0700
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
In-Reply-To: <CAMwZ-t+66=n992kX4sXwVYE6dx25iCqgLTwBLx27xgz+LMoWkw@mail.gmail.com>
References: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
	<CAMwZ-t+66=n992kX4sXwVYE6dx25iCqgLTwBLx27xgz+LMoWkw@mail.gmail.com>
Message-ID: <CAKzWOxVPgwn3kObWsgS22svKqQOWnfwvWC9y_XyYRLczPyeDKg@mail.gmail.com>

Yeah, this feature has various names, like "hot line" and "auto off-hook
dial", etc. As a previous post aptly put it, if it has a dialplan then it
probably can do a hotline. FTR I found this quickie blurb about the PAP2T
which I'm assuming will work with any Cisco/Sipura ATA FXS ports:

http://www.dslreports.com/forum/remark,16471300

Have fun and be sure to wikify this if you get it working in production.
-MC

On Tue, Sep 4, 2012 at 1:17 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Tue, Sep 4, 2012 at 1:54 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> > One of my customers has asked if we can setup a phone on an ATA (or
> > otherwise if necessary) that you just pick up and it is automatically
> > connected to some number?  Much like the ?Bat-phone? or the US to USSR
> > Emergency Phone in the Situation room out the White House.
>
> Without looking into this, I'd say you're looking at a feature on the
> SIP phone. The servers don't know if the line was picked up or not.
> So, you need hardware or software that says, "Now that the phone was
> taken off the hook, call this number." Once you have that, you can
> solve the rest of the problem in your dialplan.
>
>
> 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
>



-- 
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 Sep  5 01:06:11 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Sep 2012 14:06:11 -0700
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <6D0900A6-2881-47EF-8FEB-CC7354EA786E@vocalspace.com>
References: <CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>
	<20120904190858.cc516138@mail.tritonwest.net>
	<CAKbxfG9UL4zh1w5xTLoxqBzEHBJ94MRbY+6hvJu21Ze21ZCqsQ@mail.gmail.com>
	<6D0900A6-2881-47EF-8FEB-CC7354EA786E@vocalspace.com>
Message-ID: <CAKzWOxUrr4GiZMF65TmZu69cG+XqkiuB07S8S043K-hPQxfiYQ@mail.gmail.com>

On Tue, Sep 4, 2012 at 1:35 PM, Phillip Boles <
freeswitch-users at vocalspace.com> wrote:

> Thanks Anthony! I found the ManageSession methods by looking at the code
> that is exposed via swig.
>
> Using  "execute_on_*" hooks seems to be the proper procedure, it needs to
> be documented on the wiki. There is virtually no documentation to originate
> a call for mod managed on the wiki. I will correct that if I can get wiki
> access.
>

Visit this page:
http://wiki.freeswitch.org/wiki/Special:RequestAccount

As soon as you request wiki access then I or one of the other Wiki admins
will approve you. Once you have a wiki account then you can pretty much
edit anything you want!
-MC
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From drk at drkngs.net  Wed Sep  5 01:14:33 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 04 Sep 2012 14:14:33 -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: <6D0900A6-2881-47EF-8FEB-CC7354EA786E@vocalspace.com>
Message-ID: <20120904211433.ad092146@mail.tritonwest.net>

Yes, they should be deprecated. Mod_managed has changed a lot over the years, and a lot of those things are still left over from the days when it was either single app domain, or there were no other for controling or collecting data form outbound legs.  
   
One of the things i forgot to mention in my last reply, that is if you want to control an outbound legs completly form managed code, using the session object it is a lot easier to just do "Session.ExecuteString("originate channel/params &managed(yourclassname")". Then you can handle it just like at was an inbound call, and not even worry about clean-up.   
   
The places you may want to use the hooks are for applications where you need to originate a call from API or other context, not actually have to control the call via managed code, but need to know about the outbound leg (execute_on_originate) where you can stash UUID, and other information about the leg, or get easy notification of call being answered/terminated (api_on_answer,api_hangup_hook). With both of these API hooks, set in the "originate" api call, you can also pass variable arguments that are expanded when they are executed {origination_nested_vars=true}. Does anyone know if that is documented on the wiki? I can't find it.  
   
--Dave
      _____  

  From: Phillip Boles [mailto:freeswitch-users at vocalspace.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 04 Sep 2012 13:35:47 -0700
Subject: Re: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state

Thanks Anthony! I found the ManageSession methods by looking at the code that is exposed via swig.  

  
Using  "execute_on_*" hooks seems to be the proper procedure, it needs to be documented on the wiki. There is virtually no documentation to originate a call for mod managed on the wiki. I will correct that if I can get wiki access.  

  
I will illustrate your solution using "execute_on_originate" as the preferred solution. There are also several "execute_on_*" hooks that would work.   

  
Should ManagedSession.OriginateHandleHangup and .Originate() methods be deprecated to discourage their use in further releases? Or at least some comments in the code that point people to look at execute_on directives. Does Java have this same issue?  

  
Anthony,Glad I found a bug by doing the wrong thing.... sorry for making work for ya!  

  
I cannot say enough about how responsive Anthony and the community has been about this issue. THANKS!  

  
  
On Sep 4, 2012, at 3:10 PM, Anthony Minessale wrote:
    
Listen to Dave.  I fixed the lock leak, it was down deep into code I
am not sure is ever used.   You want to be careful with what you do
especially since you are on linux which means you must be using Mono
which I am also not sure is used a lot.



On Tue, Sep 4, 2012 at 2:08 PM, Dave R. Kompel <drk at drkngs.net> wrote:
  This may be a bug, but ManagedSession really is not the right way to do this
  from API context. Because of the APP_DOMAIN issue its much easier to just do
  Api.ExecuteString("originate ... "). If you need to get to the "managed
  session of the leg you have two options:
  
  For getting to it before the originate set the variable execute_on_originate
  to call a managed AppPlugin, or:
  
  For getting results at the end of the call in API code, set the variable
  API_HANGUP_HOOK.
  
  Both of these methoods are much easier to do from API context in
  mod_managed, and you won't have to worry about crossing app domain
  boundries, and you won't have do do any cleanup on the leg.
  
  ________________________________
  From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
  To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
  Sent: Tue, 04 Sep 2012 11:37:54 -0700
  
  Subject: Re: [Freeswitch-users] Problem with originated calls hanging in
  CS_SOFT_EXEC state
  
  now that you have a jira do not continue this thread it doubles the
  work effort, see comments there
  
  On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
  <freeswitch-users at vocalspace.com> wrote:
    I have tried to get the Current HEAD to run mod_managed with this to the
    top of switch_core.h
    #define SWITCH_DEBUG_RWLOCKS 1
    
    I am getting this error now.
    
    2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error
    Loading module /usr/local/freeswitch/mod/mod_managed.so
    **/usr/local/freeswitch/mod/mod_managed.so: undefined symbol:
    switch_core_session_read_lock**
    
    Thoughts?
    
    Thanks!
    On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:
    
      add this to the top of switch_core.h
      #define SWITCH_DEBUG_RWLOCKS 1
      
      rebuild and get a full log of the call
      look for sign of unhandled rwlock
      
      and put this on jira why I am i helping you over ml .... >=0
      
      
      On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
      <freeswitch-users at vocalspace.com> wrote:
        Sorry Yes using the latest.
        
        Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
        Author: Anthony Minessale <anthm at freeswitch.org>
        Date: Thu Aug 30 17:17:15 2012 -0500
        
        Changes made switch_cpp.cpp starting at Line 1000
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling
        CoreSession::destroy\n");
        if (session) {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have
        valid session\n");
        
        if (!channel) {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
        undefined! Trying to get it!\n");
        
        channel = switch_core_session_get_channel(session);
        }
        
        if (channel) {
        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
        "%s destroy/unlink session from object\n",
        switch_channel_get_name(channel));
        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
        "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
        switch_channel_set_private(channel, "CoreSession", NULL);
        if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) &&
        !switch_channel_test_flag(channel, CF_TRANSFER)) {
        switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
        }
        } else {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
        undefined! We Failed to get it!\n");
        }
        
        [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
        [CRIT] switch_cpp.cpp:1002 We still have valid session
        [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink
        session from object
        [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9
        destroy/unlink uuid
        [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX
        Standard REPORTING, cause: NORMAL_CLEARING
        [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX )
        State REPORTING going to sleep
        [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX )
        State Change CS_REPORTING -> CS_DESTROY
        [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX
        [BREAK]
        [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX
        ) Locked, Waiting on external entities
        
        
        
        channel remains hung
        
        P
        
        On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
        
          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
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            --
            Anthony Minessale II
            
            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire
            
            AIM: anthm
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            _________________________________________________________________________
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            _________________________________________________________________________
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            consulting at freeswitch.org
            http://www.freeswitchsolutions.com
            
            
            
            
            Official FreeSWITCH Sites
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          --
          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
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        _________________________________________________________________________
        Professional FreeSWITCH Consulting Services:
        consulting at freeswitch.org
        http://www.freeswitchsolutions.com
        
        
        
        
        Official FreeSWITCH Sites
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      --
      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
      
      
      
      
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    _________________________________________________________________________
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    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
  
  
  
  
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  http://www.freeswitchsolutions.com
  
  
  
  
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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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pstn:+19193869900

_________________________________________________________________________
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From kris at kriskinc.com  Wed Sep  5 01:29:37 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 4 Sep 2012 17:29:37 -0400
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
In-Reply-To: <CAKzWOxVPgwn3kObWsgS22svKqQOWnfwvWC9y_XyYRLczPyeDKg@mail.gmail.com>
References: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
	<CAMwZ-t+66=n992kX4sXwVYE6dx25iCqgLTwBLx27xgz+LMoWkw@mail.gmail.com>
	<CAKzWOxVPgwn3kObWsgS22svKqQOWnfwvWC9y_XyYRLczPyeDKg@mail.gmail.com>
Message-ID: <CAKDfjgc6DzwAk7LN=20nVQSeXXLgqAO3U6E15Z0bKSf+TLD3HA@mail.gmail.com>

Sipura dialplans (especially) are remarkably powerful.  Here is a
Google Books link to a few examples:

http://books.google.com/books?id=bZA2C9jLiRcC&pg=PA164&lpg=PA164&dq=voip+hacks+bat+phone&source=bl&ots=ZpgBOKtcIz&sig=wLiWdzz1sBnTmFlS0p03sFr4WN0&hl=en&sa=X&ei=e3JGUOXZKujJ0QGy5IGwDA&ved=0CDYQ6AEwAA#v=onepage&q=voip%20hacks%20bat%20phone&f=false

On Tue, Sep 4, 2012 at 5:02 PM, Michael Collins <msc at freeswitch.org> wrote:
> Yeah, this feature has various names, like "hot line" and "auto off-hook
> dial", etc. As a previous post aptly put it, if it has a dialplan then it
> probably can do a hotline. FTR I found this quickie blurb about the PAP2T
> which I'm assuming will work with any Cisco/Sipura ATA FXS ports:
>
> http://www.dslreports.com/forum/remark,16471300
>
> Have fun and be sure to wikify this if you get it working in production.
> -MC
>
>
> On Tue, Sep 4, 2012 at 1:17 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>>
>> On Tue, Sep 4, 2012 at 1:54 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> > One of my customers has asked if we can setup a phone on an ATA (or
>> > otherwise if necessary) that you just pick up and it is automatically
>> > connected to some number?  Much like the ?Bat-phone? or the US to USSR
>> > Emergency Phone in the Situation room out the White House.
>>
>> Without looking into this, I'd say you're looking at a feature on the
>> SIP phone. The servers don't know if the line was picked up or not.
>> So, you need hardware or software that says, "Now that the phone was
>> taken off the hook, call this number." Once you have that, you can
>> solve the rest of the problem in your dialplan.
>>
>>
>> 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
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>> 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
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>
> FreeSWITCH-users mailing list
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> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From all.eforums at gmail.com  Wed Sep  5 02:26:17 2012
From: all.eforums at gmail.com (A E G)
Date: Tue, 4 Sep 2012 18:26:17 -0400
Subject: [Freeswitch-users] Limitation
In-Reply-To: <CACak_yzQ4MukHSy7Hky18Ab9j29E_1Y7qO41bPOiwF5rX+6rxA@mail.gmail.com>
References: <502B8B67.3090306@pripojtese.net>
	<CAMwZ-tJdeDXKt3qEBZaqSSgkfZBqKS0+bDwugewqLGrkyc_5mg@mail.gmail.com>
	<CAJUJwtgda662nZQgx6p7Nyjq_JmzHuCwMnuxRWWv5OMe=8VnFA@mail.gmail.com>
	<CAKV-jFwcDit0hBz6Fhv9V9JWY78HxKERveWx1ZZXivbpNG_p3Q@mail.gmail.com>
	<CACak_yzQ4MukHSy7Hky18Ab9j29E_1Y7qO41bPOiwF5rX+6rxA@mail.gmail.com>
Message-ID: <CADzY+Jj6N2Oa15pY6wKLbg_PqyLFnHdYcgEJ-341O6DckLL_wA@mail.gmail.com>

On Tue, Sep 4, 2012 at 2:31 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

> .. and yes, the limitation might be changeable (easily) but won't this be
> breaking the license agreement? hint hint .. :)
>

mmm girls have told me I can't take a hint...I might not get this one
either....mind expanding on that or at least point to the license agreement
in question?
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From mike.burlingame at me.com  Wed Sep  5 02:44:28 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Tue, 04 Sep 2012 15:44:28 -0700
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
Message-ID: <F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>

I still can not compile Freeswitch with make current I am getting the following (see below) however I am able to compile the version on my box FreeSWITCH Version 1.0.head (git-b128198 2012-03-08 15-27-51 -0600) 

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 -fPI
C -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 -DSWI
TCH_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-4.0.2/libtiff -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/fre
eswitch/libs/tiff-4.0.2/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/tiff-4.0.2/libtiff/tiffio.h:33,
                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
                 from mod_spandsp.h:50,
                 from mod_spandsp.c:39:
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:68: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?int8?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:69: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?uint8?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:71: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?int16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:72: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:74: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:75: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:77: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?int64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:78: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?uint64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:94: error: expected specifier-qualifier-list before ?uint16?
cc1: warnings being treated as errors
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:96: error: struct has no members
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:98: error: expected specifier-qualifier-list before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:101: error: struct has no members
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:103: error: expected specifier-qualifier-list before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:108: error: struct has no members
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
                 from mod_spandsp.h:50,
                 from mod_spandsp.c:39:
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:67: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tmsize_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:68: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?toff_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:71: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttag_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:72: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tdir_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:73: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsample_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:74: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrile_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:75: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrip_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:76: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttile_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:77: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsize_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:145: error: expected specifier-qualifier-list before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:160: error: expected specifier-qualifier-list before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:204: error: expected specifier-qualifier-list before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:254: error: expected specifier-qualifier-list before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected declaration specifiers or ?...? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected declaration specifiers or ?...? before ?tmsize_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: type defaults to ?int? in declaration of ?tmsize_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: ?tmsize_t? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected declaration specifiers or ?...? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected declaration specifiers or ?...? before ?toff_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: type defaults to ?int? in declaration of ?toff_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: ?toff_t? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:278: error: ?TIFFSizeProc? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:285: error: parameter names (without types) in function declaration
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:286: error: expected ?)? before ?const?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:288: error: parameter names (without types) in function declaration
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:306: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetTagListEntry?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:318: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:319: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:322: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:323: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:340: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:341: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:342: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:343: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:347: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFScanlineSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:348: error: ?TIFFScanlineSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:349: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRasterScanlineSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:350: error: ?TIFFRasterScanlineSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:351: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFStripSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:352: error: ?TIFFStripSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:353: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRawStripSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error: ?TIFFRawStripSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:355: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVStripSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error: ?TIFFVStripSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:357: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileRowSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:358: error: ?TIFFTileRowSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:359: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:360: error: ?TIFFTileSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:361: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVTileSize64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error: ?TIFFVTileSize? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:363: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFDefaultStripSize?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:376: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetReadProc?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:377: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetWriteProc?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:378: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetSeekProc?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:383: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentRow?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:384: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirectory?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:385: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfDirectories?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:386: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirOffset?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:387: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentStrip?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:388: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentTile?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:398: error: expected declaration specifiers or ?...? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:399: error: expected declaration specifiers or ?...? before ?uint64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:400: error: expected declaration specifiers or ?...? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:401: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:402: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:403: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:405: error: expected declaration specifiers or ?...? before ?uint64?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected declaration specifiers or ?...? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected declaration specifiers or ?...? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected declaration specifiers or ?...? before ?TIFFReadWriteProc?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected declaration specifiers or ?...? before ?TIFFReadWriteProc?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:438: error: expected declaration specifiers or ?...? before ?TIFFSeekProc?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:452: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeTile?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected declaration specifiers or ?...? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:454: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfTiles?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected declaration specifiers or ?...? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: ?TIFFReadTile? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected declaration specifiers or ?...? before ?uint16?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: ?TIFFWriteTile? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:457: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeStrip?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:458: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfStrips?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error: ?TIFFReadEncodedStrip? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error: ?TIFFReadRawStrip? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error: ?TIFFReadEncodedTile? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error: ?TIFFReadRawTile? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error: ?TIFFWriteEncodedStrip? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error: ?TIFFWriteRawStrip? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error: ?TIFFWriteEncodedTile? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error: ?TIFFWriteRawTile? declared as function returning a function
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:469: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:470: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:471: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:474: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:475: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:476: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:477: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:480: error: expected ?)? before ?*? token
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:489: error: expected declaration specifiers or ?...? before ?uint8?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:491: error: expected ?)? before ?float?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:492: error: expected ?)? before ?float?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:503: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv24fromXYZ?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:504: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv32fromXYZ?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected declaration specifiers or ?...? before ?int32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected declaration specifiers or ?...? before ?uint32?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:525: error: expected specifier-qualifier-list before ?ttag_t?
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:533: error: struct has no members
/usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:535: error: expected declaration specifiers or ?...? before ?uint32?
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:138: error: expected specifier-qualifier-list before ?t42_encode_state_t?
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:148: 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?
/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


On Sep 4, 2012, at 1:10 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com> wrote:

> 
> Thank you mazilo and Jeff for the fix.  By looking at the GIT log I see that spandsp and libtiff are undergoing major changes these days.  I will stick to STABLE for a couple of weeks while I monitor progress on current.
>  
> 
>  
> On Tue, Sep 4, 2012 at 12:40 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
> I re-added the missing configure.gnu file.
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Missing-libtiff-after-make-current-tp7582512p7582520.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
> 
> 
> 
> -- 
> 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  Wed Sep  5 03:01:49 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Sep 2012 18:01:49 -0500
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
	<F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
Message-ID: <CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>

make spandsp-reconf

if that doesn't work you will need to manually fix libtiff

cd libs/tiff-4.0.2
autoreconf -fi
sh configure.gnu
make
cd ../..

resume as usual



On Tue, Sep 4, 2012 at 5:44 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> I still can not compile Freeswitch with make current I am getting the
> following (see below) however I am able to compile the version on my box
> FreeSWITCH Version 1.0.head (git-b128198 2012-03-08 15-27-51 -0600)
>
> 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 -fPI
> C -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 -DSWI
> TCH_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-4.0.2/libtiff
> -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/fre
> eswitch/libs/tiff-4.0.2/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/tiff-4.0.2/libtiff/tiffio.h:33,
>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
>                  from mod_spandsp.h:50,
>                  from mod_spandsp.c:39:
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:68: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?int8?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:69: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint8?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:71: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?int16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:72: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:74: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:75: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:77: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?int64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:78: error: expected ?=?,
> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:94: error: expected
> specifier-qualifier-list before ?uint16?
> cc1: warnings being treated as errors
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:96: error: struct has no
> members
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:98: error: expected
> specifier-qualifier-list before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:101: error: struct has no
> members
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:103: error: expected
> specifier-qualifier-list before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:108: error: struct has no
> members
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
>                  from mod_spandsp.h:50,
>                  from mod_spandsp.c:39:
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:67: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tmsize_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:68: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?toff_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:71: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttag_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:72: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tdir_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:73: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsample_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:74: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrile_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:75: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrip_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:76: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttile_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:77: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsize_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:145: error: expected
> specifier-qualifier-list before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:160: error: expected
> specifier-qualifier-list before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:204: error: expected
> specifier-qualifier-list before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:254: error: expected
> specifier-qualifier-list before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected
> declaration specifiers or ?...? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected
> declaration specifiers or ?...? before ?tmsize_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: type
> defaults to ?int? in declaration of ?tmsize_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: ?tmsize_t?
> declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected
> declaration specifiers or ?...? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected
> declaration specifiers or ?...? before ?toff_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: type
> defaults to ?int? in declaration of ?toff_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: ?toff_t?
> declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:278: error:
> ?TIFFSizeProc? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:285: error: parameter
> names (without types) in function declaration
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:286: error: expected
> ?)? before ?const?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:288: error: parameter
> names (without types) in function declaration
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:306: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetTagListEntry?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:318: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:319: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:322: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:323: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:340: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:341: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:342: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:343: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:347: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFScanlineSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:348: error:
> ?TIFFScanlineSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:349: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRasterScanlineSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:350: error:
> ?TIFFRasterScanlineSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:351: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFStripSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:352: error:
> ?TIFFStripSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:353: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRawStripSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error:
> ?TIFFRawStripSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:355: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVStripSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error:
> ?TIFFVStripSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:357: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileRowSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:358: error:
> ?TIFFTileRowSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:359: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:360: error:
> ?TIFFTileSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:361: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVTileSize64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error:
> ?TIFFVTileSize? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:363: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFDefaultStripSize?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:376: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetReadProc?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:377: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetWriteProc?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:378: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetSeekProc?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:383: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentRow?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:384: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirectory?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:385: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfDirectories?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:386: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirOffset?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:387: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentStrip?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:388: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentTile?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:398: error: expected
> declaration specifiers or ?...? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:399: error: expected
> declaration specifiers or ?...? before ?uint64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:400: error: expected
> declaration specifiers or ?...? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:401: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:402: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:403: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:405: error: expected
> declaration specifiers or ?...? before ?uint64?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected
> declaration specifiers or ?...? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected
> declaration specifiers or ?...? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected
> declaration specifiers or ?...? before ?TIFFReadWriteProc?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected
> declaration specifiers or ?...? before ?TIFFReadWriteProc?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:438: error: expected
> declaration specifiers or ?...? before ?TIFFSeekProc?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:452: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeTile?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
> declaration specifiers or ?...? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:454: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfTiles?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
> declaration specifiers or ?...? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error:
> ?TIFFReadTile? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
> declaration specifiers or ?...? before ?uint16?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error:
> ?TIFFWriteTile? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:457: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeStrip?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:458: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfStrips?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error:
> ?TIFFReadEncodedStrip? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error:
> ?TIFFReadRawStrip? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error:
> ?TIFFReadEncodedTile? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error:
> ?TIFFReadRawTile? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error:
> ?TIFFWriteEncodedStrip? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error:
> ?TIFFWriteRawStrip? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error:
> ?TIFFWriteEncodedTile? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error:
> ?TIFFWriteRawTile? declared as function returning a function
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:469: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:470: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:471: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:474: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:475: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:476: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:477: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:480: error: expected
> ?)? before ?*? token
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:489: error: expected
> declaration specifiers or ?...? before ?uint8?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:491: error: expected
> ?)? before ?float?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:492: error: expected
> ?)? before ?float?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:503: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv24fromXYZ?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:504: error: expected
> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv32fromXYZ?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
> declaration specifiers or ?...? before ?int32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
> declaration specifiers or ?...? before ?uint32?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:525: error: expected
> specifier-qualifier-list before ?ttag_t?
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:533: error: struct has
> no members
> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:535: error: expected
> declaration specifiers or ?...? before ?uint32?
> 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:138: error:
> expected specifier-qualifier-list before ?t42_encode_state_t?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:148: 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?
> /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
>
>
> On Sep 4, 2012, at 1:10 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com>
> wrote:
>
>
> Thank you mazilo and Jeff for the fix.  By looking at the GIT log I see that
> spandsp and libtiff are undergoing major changes these days.  I will stick
> to STABLE for a couple of weeks while I monitor progress on current.
>
>
>
> On Tue, Sep 4, 2012 at 12:40 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>
>> I re-added the missing configure.gnu file.
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Missing-libtiff-after-make-current-tp7582512p7582520.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
>
>
>
>
> --
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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FreeSWITCH Developer Conference
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pstn:+19193869900


From mike.burlingame at me.com  Wed Sep  5 03:15:15 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Tue, 04 Sep 2012 16:15:15 -0700
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
	<F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
	<CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>
Message-ID: <95BF6B32-E362-4882-BA1C-F228058B4442@me.com>

thanks that seem to get me past that point now I have another issue I will deal with later and start a new topic if I can not resolve seeing it's no longer related

kernel: [4076389.400073] freeswitch[13231]: segfault at 0 ip 00007f328d1cefa6 sp 00007fff2464d5d0 error 4 in libfreeswitch.so.1.0.0[7f328d13a000+210000]


On Sep 4, 2012, at 4:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> make spandsp-reconf
> 
> if that doesn't work you will need to manually fix libtiff
> 
> cd libs/tiff-4.0.2
> autoreconf -fi
> sh configure.gnu
> make
> cd ../..
> 
> resume as usual
> 
> 
> 
> On Tue, Sep 4, 2012 at 5:44 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> I still can not compile Freeswitch with make current I am getting the
>> following (see below) however I am able to compile the version on my box
>> FreeSWITCH Version 1.0.head (git-b128198 2012-03-08 15-27-51 -0600)
>> 
>> 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 -fPI
>> C -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 -DSWI
>> TCH_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-4.0.2/libtiff
>> -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/fre
>> eswitch/libs/tiff-4.0.2/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/tiff-4.0.2/libtiff/tiffio.h:33,
>>                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
>>                 from mod_spandsp.h:50,
>>                 from mod_spandsp.c:39:
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:68: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int8?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:69: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint8?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:71: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:72: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:74: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:75: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:77: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:78: error: expected ?=?,
>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:94: error: expected
>> specifier-qualifier-list before ?uint16?
>> cc1: warnings being treated as errors
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:96: error: struct has no
>> members
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:98: error: expected
>> specifier-qualifier-list before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:101: error: struct has no
>> members
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:103: error: expected
>> specifier-qualifier-list before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:108: error: struct has no
>> members
>> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
>>                 from mod_spandsp.h:50,
>>                 from mod_spandsp.c:39:
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:67: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tmsize_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:68: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?toff_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:71: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttag_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:72: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tdir_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:73: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsample_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:74: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrile_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:75: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrip_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:76: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttile_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:77: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsize_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:145: error: expected
>> specifier-qualifier-list before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:160: error: expected
>> specifier-qualifier-list before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:204: error: expected
>> specifier-qualifier-list before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:254: error: expected
>> specifier-qualifier-list before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected
>> declaration specifiers or ?...? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected
>> declaration specifiers or ?...? before ?tmsize_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: type
>> defaults to ?int? in declaration of ?tmsize_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: ?tmsize_t?
>> declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected
>> declaration specifiers or ?...? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected
>> declaration specifiers or ?...? before ?toff_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: type
>> defaults to ?int? in declaration of ?toff_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: ?toff_t?
>> declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:278: error:
>> ?TIFFSizeProc? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:285: error: parameter
>> names (without types) in function declaration
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:286: error: expected
>> ?)? before ?const?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:288: error: parameter
>> names (without types) in function declaration
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:306: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetTagListEntry?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:318: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:319: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:322: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:323: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:340: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:341: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:342: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:343: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:347: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFScanlineSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:348: error:
>> ?TIFFScanlineSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:349: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRasterScanlineSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:350: error:
>> ?TIFFRasterScanlineSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:351: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFStripSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:352: error:
>> ?TIFFStripSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:353: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRawStripSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error:
>> ?TIFFRawStripSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:355: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVStripSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error:
>> ?TIFFVStripSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:357: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileRowSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:358: error:
>> ?TIFFTileRowSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:359: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:360: error:
>> ?TIFFTileSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:361: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVTileSize64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error:
>> ?TIFFVTileSize? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:363: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFDefaultStripSize?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:376: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetReadProc?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:377: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetWriteProc?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:378: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetSeekProc?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:383: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentRow?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:384: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirectory?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:385: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfDirectories?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:386: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirOffset?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:387: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentStrip?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:388: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentTile?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:398: error: expected
>> declaration specifiers or ?...? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:399: error: expected
>> declaration specifiers or ?...? before ?uint64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:400: error: expected
>> declaration specifiers or ?...? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:401: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:402: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:403: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:405: error: expected
>> declaration specifiers or ?...? before ?uint64?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected
>> declaration specifiers or ?...? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected
>> declaration specifiers or ?...? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected
>> declaration specifiers or ?...? before ?TIFFReadWriteProc?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected
>> declaration specifiers or ?...? before ?TIFFReadWriteProc?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:438: error: expected
>> declaration specifiers or ?...? before ?TIFFSeekProc?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:452: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeTile?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>> declaration specifiers or ?...? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:454: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfTiles?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>> declaration specifiers or ?...? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error:
>> ?TIFFReadTile? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>> declaration specifiers or ?...? before ?uint16?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error:
>> ?TIFFWriteTile? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:457: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeStrip?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:458: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfStrips?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error:
>> ?TIFFReadEncodedStrip? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error:
>> ?TIFFReadRawStrip? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error:
>> ?TIFFReadEncodedTile? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error:
>> ?TIFFReadRawTile? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error:
>> ?TIFFWriteEncodedStrip? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error:
>> ?TIFFWriteRawStrip? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error:
>> ?TIFFWriteEncodedTile? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error:
>> ?TIFFWriteRawTile? declared as function returning a function
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:469: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:470: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:471: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:474: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:475: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:476: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:477: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:480: error: expected
>> ?)? before ?*? token
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:489: error: expected
>> declaration specifiers or ?...? before ?uint8?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:491: error: expected
>> ?)? before ?float?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:492: error: expected
>> ?)? before ?float?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:503: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv24fromXYZ?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:504: error: expected
>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv32fromXYZ?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
>> declaration specifiers or ?...? before ?int32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:525: error: expected
>> specifier-qualifier-list before ?ttag_t?
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:533: error: struct has
>> no members
>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:535: error: expected
>> declaration specifiers or ?...? before ?uint32?
>> 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:138: error:
>> expected specifier-qualifier-list before ?t42_encode_state_t?
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:148: 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?
>> /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
>> 
>> 
>> On Sep 4, 2012, at 1:10 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com>
>> wrote:
>> 
>> 
>> Thank you mazilo and Jeff for the fix.  By looking at the GIT log I see that
>> spandsp and libtiff are undergoing major changes these days.  I will stick
>> to STABLE for a couple of weeks while I monitor progress on current.
>> 
>> 
>> 
>> On Tue, Sep 4, 2012 at 12:40 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>> 
>>> I re-added the missing configure.gnu file.
>>> 
>>> 
>>> 
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Missing-libtiff-after-make-current-tp7582512p7582520.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
>> 
>> 
>> 
>> 
>> --
>> 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
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
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> 
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> 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 anthony.minessale at gmail.com  Wed Sep  5 03:18:07 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Sep 2012 18:18:07 -0500
Subject: [Freeswitch-users] Missing libtiff after make current
In-Reply-To: <95BF6B32-E362-4882-BA1C-F228058B4442@me.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
	<F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
	<CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>
	<95BF6B32-E362-4882-BA1C-F228058B4442@me.com>
Message-ID: <CAKbxfG9cLhx-eTRx28WfVf-XEKcey+LGeEGsJbsqfFFoE4xqKw@mail.gmail.com>

for that one see

http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace


On Tue, Sep 4, 2012 at 6:15 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> thanks that seem to get me past that point now I have another issue I will deal with later and start a new topic if I can not resolve seeing it's no longer related
>
> kernel: [4076389.400073] freeswitch[13231]: segfault at 0 ip 00007f328d1cefa6 sp 00007fff2464d5d0 error 4 in libfreeswitch.so.1.0.0[7f328d13a000+210000]
>
>
> On Sep 4, 2012, at 4:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> make spandsp-reconf
>>
>> if that doesn't work you will need to manually fix libtiff
>>
>> cd libs/tiff-4.0.2
>> autoreconf -fi
>> sh configure.gnu
>> make
>> cd ../..
>>
>> resume as usual
>>
>>
>>
>> On Tue, Sep 4, 2012 at 5:44 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> I still can not compile Freeswitch with make current I am getting the
>>> following (see below) however I am able to compile the version on my box
>>> FreeSWITCH Version 1.0.head (git-b128198 2012-03-08 15-27-51 -0600)
>>>
>>> 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 -fPI
>>> C -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 -DSWI
>>> TCH_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-4.0.2/libtiff
>>> -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/fre
>>> eswitch/libs/tiff-4.0.2/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/tiff-4.0.2/libtiff/tiffio.h:33,
>>>                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
>>>                 from mod_spandsp.h:50,
>>>                 from mod_spandsp.c:39:
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:68: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int8?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:69: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint8?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:71: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:72: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:74: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:75: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:77: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?int64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:78: error: expected ?=?,
>>> ?,?, ?;?, ?asm? or ?__attribute__? before ?uint64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:94: error: expected
>>> specifier-qualifier-list before ?uint16?
>>> cc1: warnings being treated as errors
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:96: error: struct has no
>>> members
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:98: error: expected
>>> specifier-qualifier-list before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:101: error: struct has no
>>> members
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:103: error: expected
>>> specifier-qualifier-list before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiff.h:108: error: struct has no
>>> members
>>> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:43,
>>>                 from mod_spandsp.h:50,
>>>                 from mod_spandsp.c:39:
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:67: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tmsize_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:68: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?toff_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:71: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttag_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:72: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tdir_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:73: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsample_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:74: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrile_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:75: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tstrip_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:76: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?ttile_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:77: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?tsize_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:145: error: expected
>>> specifier-qualifier-list before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:160: error: expected
>>> specifier-qualifier-list before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:191: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:194: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:204: error: expected
>>> specifier-qualifier-list before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:254: error: expected
>>> specifier-qualifier-list before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected
>>> declaration specifiers or ?...? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: expected
>>> declaration specifiers or ?...? before ?tmsize_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: type
>>> defaults to ?int? in declaration of ?tmsize_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:275: error: ?tmsize_t?
>>> declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected
>>> declaration specifiers or ?...? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: expected
>>> declaration specifiers or ?...? before ?toff_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: type
>>> defaults to ?int? in declaration of ?toff_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:276: error: ?toff_t?
>>> declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:278: error:
>>> ?TIFFSizeProc? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:285: error: parameter
>>> names (without types) in function declaration
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:286: error: expected
>>> ?)? before ?const?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:288: error: parameter
>>> names (without types) in function declaration
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:306: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetTagListEntry?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:318: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:319: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:322: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:323: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:340: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:341: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:342: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:343: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:347: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFScanlineSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:348: error:
>>> ?TIFFScanlineSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:349: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRasterScanlineSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:350: error:
>>> ?TIFFRasterScanlineSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:351: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFStripSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:352: error:
>>> ?TIFFStripSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:353: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFRawStripSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:354: error:
>>> ?TIFFRawStripSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:355: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVStripSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:356: error:
>>> ?TIFFVStripSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:357: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileRowSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:358: error:
>>> ?TIFFTileRowSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:359: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFTileSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:360: error:
>>> ?TIFFTileSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:361: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFVTileSize64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:362: error:
>>> ?TIFFVTileSize? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:363: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFDefaultStripSize?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:364: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:376: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetReadProc?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:377: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetWriteProc?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:378: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFGetSeekProc?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:383: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentRow?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:384: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirectory?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:385: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfDirectories?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:386: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentDirOffset?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:387: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentStrip?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:388: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFCurrentTile?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:398: error: expected
>>> declaration specifiers or ?...? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:399: error: expected
>>> declaration specifiers or ?...? before ?uint64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:400: error: expected
>>> declaration specifiers or ?...? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:401: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:402: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:403: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:405: error: expected
>>> declaration specifiers or ?...? before ?uint64?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:418: error: expected
>>> declaration specifiers or ?...? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:419: error: expected
>>> declaration specifiers or ?...? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:420: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:421: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:424: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:425: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:428: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected
>>> declaration specifiers or ?...? before ?TIFFReadWriteProc?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:437: error: expected
>>> declaration specifiers or ?...? before ?TIFFReadWriteProc?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:438: error: expected
>>> declaration specifiers or ?...? before ?TIFFSeekProc?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:452: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeTile?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:453: error: expected
>>> declaration specifiers or ?...? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:454: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfTiles?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error: expected
>>> declaration specifiers or ?...? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:455: error:
>>> ?TIFFReadTile? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error: expected
>>> declaration specifiers or ?...? before ?uint16?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:456: error:
>>> ?TIFFWriteTile? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:457: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFComputeStrip?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:458: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?TIFFNumberOfStrips?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:459: error:
>>> ?TIFFReadEncodedStrip? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:460: error:
>>> ?TIFFReadRawStrip? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:461: error:
>>> ?TIFFReadEncodedTile? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:462: error:
>>> ?TIFFReadRawTile? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:463: error:
>>> ?TIFFWriteEncodedStrip? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:464: error:
>>> ?TIFFWriteRawStrip? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:465: error:
>>> ?TIFFWriteEncodedTile? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:466: error:
>>> ?TIFFWriteRawTile? declared as function returning a function
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:469: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:470: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:471: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:474: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:475: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:476: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:477: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:480: error: expected
>>> ?)? before ?*? token
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:489: error: expected
>>> declaration specifiers or ?...? before ?uint8?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:491: error: expected
>>> ?)? before ?float?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:492: error: expected
>>> ?)? before ?float?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:503: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv24fromXYZ?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:504: error: expected
>>> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?LogLuv32fromXYZ?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:509: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:512: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:515: error: expected
>>> declaration specifiers or ?...? before ?int32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:516: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:525: error: expected
>>> specifier-qualifier-list before ?ttag_t?
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:533: error: struct has
>>> no members
>>> /usr/src/freeswitch/libs/tiff-4.0.2/libtiff/tiffio.h:535: error: expected
>>> declaration specifiers or ?...? before ?uint32?
>>> 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:138: error:
>>> expected specifier-qualifier-list before ?t42_encode_state_t?
>>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:148: 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?
>>> /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
>>>
>>>
>>> On Sep 4, 2012, at 1:10 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com>
>>> wrote:
>>>
>>>
>>> Thank you mazilo and Jeff for the fix.  By looking at the GIT log I see that
>>> spandsp and libtiff are undergoing major changes these days.  I will stick
>>> to STABLE for a couple of weeks while I monitor progress on current.
>>>
>>>
>>>
>>> On Tue, Sep 4, 2012 at 12:40 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>>>
>>>> I re-added the missing configure.gnu file.
>>>>
>>>>
>>>>
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/Missing-libtiff-after-make-current-tp7582512p7582520.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
>>>
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 daveh at beachdognet.com  Wed Sep  5 06:33:04 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Tue, 4 Sep 2012 22:33:04 -0400
Subject: [Freeswitch-users] problem with bridged conference
Message-ID: <k26dl0$iou$1@ger.gmane.org>

I am having trouble getting a simple bridged conference to work.  The 
scenario is simply that an arriving call is answered, and I then want 
to create a bridged conference and outdial a second participant.  I am 
trying to do that with the following dialplan:

    <extension name="6174973896" continue="false">
      <condition field="source" expression="mod_sofia"/>
      <condition field="destination_number" expression="6174973896">
                <action application="ring_ready"/>
                <action application="answer"/>
                <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>

However, this is not working.  The designated number is outdialed; 
however there is no audio connection between the caller and called 
party.  Also, when either party hangs up, the other party remains in 
the conference, even though a bridged conference is supposed to be torn 
down when the number of parties drops to 1.

I'm sure it's something simple, but need some help.  I have a pastebin 
of the debug logging when I run the scenario at 
http://pastebin.freeswitch.org/19838

Thanks

Dave




From gabe at gundy.org  Wed Sep  5 06:41:30 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 4 Sep 2012 20:41:30 -0600
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
In-Reply-To: <CAKDfjgc6DzwAk7LN=20nVQSeXXLgqAO3U6E15Z0bKSf+TLD3HA@mail.gmail.com>
References: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
	<CAMwZ-t+66=n992kX4sXwVYE6dx25iCqgLTwBLx27xgz+LMoWkw@mail.gmail.com>
	<CAKzWOxVPgwn3kObWsgS22svKqQOWnfwvWC9y_XyYRLczPyeDKg@mail.gmail.com>
	<CAKDfjgc6DzwAk7LN=20nVQSeXXLgqAO3U6E15Z0bKSf+TLD3HA@mail.gmail.com>
Message-ID: <CAMwZ-tLHNSGuz07y8jGA1ZmjUda7CfifF+_tQPEOoUNVM=z22Q@mail.gmail.com>

On Tue, Sep 4, 2012 at 3:29 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Sipura dialplans (especially) are remarkably powerful.  Here is a
> Google Books link to a few examples:


Heh, I have that book, but I've never picked it up; maybe I should.

Can you vouch for any of the authors? ;)


Gabe


From mike.burlingame at me.com  Wed Sep  5 07:50:43 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Tue, 04 Sep 2012 20:50:43 -0700
Subject: [Freeswitch-users] Segfault from make current
In-Reply-To: <CAKbxfG9cLhx-eTRx28WfVf-XEKcey+LGeEGsJbsqfFFoE4xqKw@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
	<F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
	<CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>
	<95BF6B32-E362-4882-BA1C-F228058B4442@me.com>
	<CAKbxfG9cLhx-eTRx28WfVf-XEKcey+LGeEGsJbsqfFFoE4xqKw@mail.gmail.com>
Message-ID: <7CD171F0-6BF1-4203-8FBF-93B6FE326E34@me.com>

Branching out from the libtiff string core dump info posted to http://pastebin.freeswitch.org/19839


On Sep 4, 2012, at 4:18 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> for that one see
> 
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace
> 
> 
> On Tue, Sep 4, 2012 at 6:15 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> thanks that seem to get me past that point now I have another issue I will deal with later and start a new topic if I can not resolve seeing it's no longer related
>> 
>> kernel: [4076389.400073] freeswitch[13231]: segfault at 0 ip 00007f328d1cefa6 sp 00007fff2464d5d0 error 4 in libfreeswitch.so.1.0.0[7f328d13a000+210000]
>> 
>> 
>> On Sep 4, 2012, at 4:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> make spandsp-reconf
>>> 
>>> if that doesn't work you will need to manually fix libtiff
>>> 
>>> cd libs/tiff-4.0.2
>>> autoreconf -fi
>>> sh configure.gnu
>>> make
>>> cd ../..
>>> 
>>> resume as usual
>>> 
>>> 
>>> 



From curriegrad2004 at gmail.com  Wed Sep  5 08:13:50 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 4 Sep 2012 21:13:50 -0700
Subject: [Freeswitch-users] Segfault from make current
In-Reply-To: <7CD171F0-6BF1-4203-8FBF-93B6FE326E34@me.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
	<F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
	<CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>
	<95BF6B32-E362-4882-BA1C-F228058B4442@me.com>
	<CAKbxfG9cLhx-eTRx28WfVf-XEKcey+LGeEGsJbsqfFFoE4xqKw@mail.gmail.com>
	<7CD171F0-6BF1-4203-8FBF-93B6FE326E34@me.com>
Message-ID: <CAMWiMy9_wL0pFj79EPpODRwF7F2OxrHgowkZjvqYSV2b049SLg@mail.gmail.com>

Please report this as a bug on JIRA.

On Tue, Sep 4, 2012 at 8:50 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> Branching out from the libtiff string core dump info posted to http://pastebin.freeswitch.org/19839
>
>
> On Sep 4, 2012, at 4:18 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> for that one see
>>
>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace
>>
>>
>> On Tue, Sep 4, 2012 at 6:15 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> thanks that seem to get me past that point now I have another issue I will deal with later and start a new topic if I can not resolve seeing it's no longer related
>>>
>>> kernel: [4076389.400073] freeswitch[13231]: segfault at 0 ip 00007f328d1cefa6 sp 00007fff2464d5d0 error 4 in libfreeswitch.so.1.0.0[7f328d13a000+210000]
>>>
>>>
>>> On Sep 4, 2012, at 4:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>
>>>> make spandsp-reconf
>>>>
>>>> if that doesn't work you will need to manually fix libtiff
>>>>
>>>> cd libs/tiff-4.0.2
>>>> autoreconf -fi
>>>> sh configure.gnu
>>>> make
>>>> cd ../..
>>>>
>>>> resume as usual
>>>>
>>>>
>>>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Wed Sep  5 10:14:29 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 05 Sep 2012 09:14:29 +0300
Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file
Message-ID: <5046EDC5.8080605@gmail.com>

Hello.

Regularly in FS log I see:

09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]: Storing ZRTP 
Cache...
09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]:    Storing 
ZRTP cache to <...
09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]:    ERROR! 
unable to open ZRTP cache file <.
09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]: Storing ZRTP 
Cache - DONE.
09:03:54.044779 [DEBUG] switch_rtp.c:776 Saving ZRTP cache: OK

Why is this logged, and why an error? There is no file name to search 
for permission or access. Note the missing ">" where file name must be.

-- 
Mimiko desu.


From steveayre at gmail.com  Wed Sep  5 12:39:21 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 5 Sep 2012 09:39:21 +0100
Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file
In-Reply-To: <5046EDC5.8080605@gmail.com>
References: <5046EDC5.8080605@gmail.com>
Message-ID: <CAFiqYuk-g+xxQ2g8cGo08q2_mgHvj8hkD_NPpCy58zHHZqKEiQ@mail.gmail.com>

It's a log message from within the libzrtp library.

It means it cannot open the cache file. The log message contains the
filename it tried to open, but it looks like you've removed the end of
the lines with that information.
The default filename within the library is
"./zrtp_def_cache_path.dat". That'd be within the cwd FS is running
with, which you can find with "ls -ld /proc/PID/cwd" where PID is the
PID freeswitch is running as.

The most likely cause is perhaps a write permission error. libzrtp
doesn't log the error that occurred, but you might be able to see it
by attaching strace, but it'd first be easier to check if FS can write
to its cwd directory.

-Steve




On 5 September 2012 07:14, Vbvbrj <vbvbrj at gmail.com> wrote:
> Hello.
>
> Regularly in FS log I see:
>
> 09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]: Storing ZRTP
> Cache...
> 09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]:    Storing
> ZRTP cache to <...
> 09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]:    ERROR!
> unable to open ZRTP cache file <.
> 09:03:54.044779 [DEBUG] switch_rtp.c:915  [  zrtp cache]: Storing ZRTP
> Cache - DONE.
> 09:03:54.044779 [DEBUG] switch_rtp.c:776 Saving ZRTP cache: OK
>
> Why is this logged, and why an error? There is no file name to search
> for permission or access. Note the missing ">" where file name must be.
>
> --
> 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


From steveayre at gmail.com  Wed Sep  5 12:42:22 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 5 Sep 2012 09:42:22 +0100
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
In-Reply-To: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
References: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
Message-ID: <CAFiqYum8tgGJHPdPgczeTb=S1zaVj9hyLXZCG1CCG0PLz-gdcw@mail.gmail.com>

Do you have SIP traces for the call? If you're bridging then it's
possible for the remote end to hangup the answered call with an ISDN
clearing cause other than NORMAL_* (just like they can in ISDN
signalling). For billing purposes I suggest you bill anything where
billsec>0 rather than only NORMAL_CLEARING calls.

-Steve



On 4 September 2012 19:23, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
> Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
> NORMAL_CLEARING while the calls still have a valid duration and answered
> time. For a typical billing scenario, calls are billed if (answered time)
> billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the enumeration
> would be other than normal and while the calls would be still a valid
> answered call?
>
> 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  Wed Sep  5 14:19:33 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 05 Sep 2012 13:19:33 +0300
Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file
In-Reply-To: <CAFiqYuk-g+xxQ2g8cGo08q2_mgHvj8hkD_NPpCy58zHHZqKEiQ@mail.gmail.com>
References: <5046EDC5.8080605@gmail.com>
	<CAFiqYuk-g+xxQ2g8cGo08q2_mgHvj8hkD_NPpCy58zHHZqKEiQ@mail.gmail.com>
Message-ID: <50472735.7080601@gmail.com>

On 05.09.2012 11:39, Steven Ayre wrote:
> It's a log message from within the libzrtp library.
>
> It means it cannot open the cache file. The log message contains the
> filename it tried to open, but it looks like you've removed the end of
> the lines with that information.

No, I didn't remove any symbol from the log output. Realy it does not 
show the file name.

> The default filename within the library is
> "./zrtp_def_cache_path.dat". That'd be within the cwd FS is running
> with, which you can find with "ls -ld /proc/PID/cwd" where PID is the
> PID freeswitch is running as.

Yes, the cmd of FS is running is correct and the username under which FS 
is running does not have write permission except some subfolders, for 
ex. log sobfolder. Then how to specify the zrtp cache file path and name?

-- 
Mimiko desu.


From yehavi.bourvine at gmail.com  Wed Sep  5 14:56:51 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 5 Sep 2012 13:56:51 +0300
Subject: [Freeswitch-users] effective_callee_id_name behaviour
Message-ID: <CAAtWKeEXaWLfXL89QcwxHCHXWE07up13mgOwUUCytRNpv9EogQ@mail.gmail.com>

Hello,

  Our FreeSwitch is connected to a Nortel PBX via E1-SIP gateway
(AudioCodes). This gateway supports the P-Asserted-ID field.

With a vanilla configuration the following happens when a FS user calls a
user on the Nortel:
- During the ringing phase the name of the callee is "outbound call"
- When the other side answers the name of the callee is set to the one sent
from the Nortel.

In order to have the name also during the ringing phase, I set
calle_id_name and effective_callee_id_name.
After doing so, the name is not changed after the remote user answers. I
need it to change to the name sent
from the Nortel, as it shows the name of the one who actually answered the
phone.

I also tried adding ignore_display_updates=false, but it didn't change the
behaviour.

Any idea how to do it?

                        Thanks! __Yehavi:
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From daveh at beachdognet.com  Wed Sep  5 17:15:55 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Wed, 5 Sep 2012 09:15:55 -0400
Subject: [Freeswitch-users] problem with bridged conference
References: <k26dl0$iou$1@ger.gmane.org>
Message-ID: <k27ja8$ua3$1@ger.gmane.org>

This appears to be a bug having to do with the use of the literal 
_uuid_ for the conference name.  The docs say (and the code agrees, 
from what I can see) that this literal will cause the conference name 
to be set to the uuid of channel; i.e., a guaranteed unique conference 
name.  However, looking further at the logs I posted before, it seems 
that what is happening when I use that is that two conferences are 
actually being created -- one for the original caller and one for the 
called party.  You can see, for instance, that two different set of 
caller controls are established, which indicates that two conference 
threads have been spun up, and two conference rooms created.  Thus 
caller and called party can't hear each other.

I changed the conference name from _uuid_ to a specific name of my 
choosing (also guaranteed to be unique), and things appear to be 
working properly.

I am running a fairly old version of freeswitch (FreeSWITCH Version 
1.0.head (git-426a448 2010-09-27 10-12-10 -0400) so perhaps this is 
something that has been fixed in a more recent version (?)

Dave

On 2012-09-05 02:33:04 +0000, Dave Horton said:

> I am having trouble getting a simple bridged conference to work.  The
> scenario is simply that an arriving call is answered, and I then want
> to create a bridged conference and outdial a second participant.  I am
> trying to do that with the following dialplan:
> 
>     <extension name="6174973896" continue="false">
>       <condition field="source" expression="mod_sofia"/>
>       <condition field="destination_number" expression="6174973896">
>                 <action application="ring_ready"/>
>                 <action application="answer"/>
>                 <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>
> 
> However, this is not working.  The designated number is outdialed;
> however there is no audio connection between the caller and called
> party.  Also, when either party hangs up, the other party remains in
> the conference, even though a bridged conference is supposed to be torn
> down when the number of parties drops to 1.
> 
> I'm sure it's something simple, but need some help.  I have a pastebin
> of the debug logging when I run the scenario at
> http://pastebin.freeswitch.org/19838





From freeswitch-users at vocalspace.com  Wed Sep  5 01:51:07 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Tue, 4 Sep 2012 16:51:07 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging
	in	CS_SOFT_EXEC state
In-Reply-To: <20120904211433.ad092146@mail.tritonwest.net>
References: <20120904211433.ad092146@mail.tritonwest.net>
Message-ID: <AC7BA137-528C-4292-980C-5A42650A340B@vocalspace.com>

I see. It would seem since the whole project compiles in VS2008 & VS2010 that there would be more mod_managed users.

Javascript/Lua seem to be the front runners for scripting and in general they should be the choice. Our particular spec requires real time interaction with web servers using socket.io. Neither of these languages made it easy to watch calls except through ESL knowing the UUID. We wanted to throw events directly from the code to a Message Queue server and deal only with what we want.

Mono/C# got selected since we already have a very extensive custom module built on it by another developer in our shop. I will try to write docs for the project since I will be doing more than the average bear with mod_managed. Any doc ideas you or anyone else has shoot them over and I will try to make examples and docs. I also would appreciate if you would not mind me hitting you up to proof read my wiki entries for accuracy once they are done.

Phillip

On Sep 4, 2012, at 4:14 PM, Dave R. Kompel wrote:

> Yes, they should be deprecated. Mod_managed has changed a lot over the years, and a lot of those things are still left over from the days when it was either single app domain, or there were no other for controling or collecting data form outbound legs.
>  
> One of the things i forgot to mention in my last reply, that is if you want to control an outbound legs completly form managed code, using the session object it is a lot easier to just do "Session.ExecuteString("originate channel/params &managed(yourclassname")". Then you can handle it just like at was an inbound call, and not even worry about clean-up.
>  
> The places you may want to use the hooks are for applications where you need to originate a call from API or other context, not actually have to control the call via managed code, but need to know about the outbound leg (execute_on_originate) where you can stash UUID, and other information about the leg, or get easy notification of call being answered/terminated (api_on_answer,api_hangup_hook). With both of these API hooks, set in the "originate" api call, you can also pass variable arguments that are expanded when they are executed {origination_nested_vars=true}. Does anyone know if that is documented on the wiki? I can't find it.
>  
> --Dave
> 
> From: Phillip Boles [mailto:freeswitch-users at vocalspace.com]
> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Tue, 04 Sep 2012 13:35:47 -0700
> Subject: Re: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state
> 
> Thanks Anthony! I found the ManageSession methods by looking at the code that is exposed via swig.
> 
> Using  "execute_on_*" hooks seems to be the proper procedure, it needs to be documented on the wiki. There is virtually no documentation to originate a call for mod managed on the wiki. I will correct that if I can get wiki access.
> 
> I will illustrate your solution using "execute_on_originate" as the preferred solution. There are also several "execute_on_*" hooks that would work. 
> 
> Should ManagedSession.OriginateHandleHangup and .Originate() methods be deprecated to discourage their use in further releases? Or at least some comments in the code that point people to look at execute_on directives. Does Java have this same issue?
> 
> Anthony,Glad I found a bug by doing the wrong thing.... sorry for making work for ya!
> 
> I cannot say enough about how responsive Anthony and the community has been about this issue. THANKS!
> 
> On Sep 4, 2012, at 3:10 PM, Anthony Minessale wrote:
> 
>> Listen to Dave.  I fixed the lock leak, it was down deep into code I
>> am not sure is ever used.   You want to be careful with what you do
>> especially since you are on linux which means you must be using Mono
>> which I am also not sure is used a lot.
>> 
>> 
>> 
>> On Tue, Sep 4, 2012 at 2:08 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>> This may be a bug, but ManagedSession really is not the right way to do this
>>> from API context. Because of the APP_DOMAIN issue its much easier to just do
>>> Api.ExecuteString("originate ... "). If you need to get to the "managed
>>> session of the leg you have two options:
>>> 
>>> For getting to it before the originate set the variable execute_on_originate
>>> to call a managed AppPlugin, or:
>>> 
>>> For getting results at the end of the call in API code, set the variable
>>> API_HANGUP_HOOK.
>>> 
>>> Both of these methoods are much easier to do from API context in
>>> mod_managed, and you won't have to worry about crossing app domain
>>> boundries, and you won't have do do any cleanup on the leg.
>>> 
>>> ________________________________
>>> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
>>> Sent: Tue, 04 Sep 2012 11:37:54 -0700
>>> 
>>> Subject: Re: [Freeswitch-users] Problem with originated calls hanging in
>>> CS_SOFT_EXEC state
>>> 
>>> now that you have a jira do not continue this thread it doubles the
>>> work effort, see comments there
>>> 
>>> On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
>>> <freeswitch-users at vocalspace.com> wrote:
>>>> I have tried to get the Current HEAD to run mod_managed with this to the
>>>> top of switch_core.h
>>>> #define SWITCH_DEBUG_RWLOCKS 1
>>>> 
>>>> I am getting this error now.
>>>> 
>>>> 2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error
>>>> Loading module /usr/local/freeswitch/mod/mod_managed.so
>>>> **/usr/local/freeswitch/mod/mod_managed.so: undefined symbol:
>>>> switch_core_session_read_lock**
>>>> 
>>>> Thoughts?
>>>> 
>>>> Thanks!
>>>> On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:
>>>> 
>>>>> add this to the top of switch_core.h
>>>>> #define SWITCH_DEBUG_RWLOCKS 1
>>>>> 
>>>>> rebuild and get a full log of the call
>>>>> look for sign of unhandled rwlock
>>>>> 
>>>>> and put this on jira why I am i helping you over ml .... >=0
>>>>> 
>>>>> 
>>>>> On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
>>>>> <freeswitch-users at vocalspace.com> wrote:
>>>>>> Sorry Yes using the latest.
>>>>>> 
>>>>>> Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
>>>>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>>>> Date: Thu Aug 30 17:17:15 2012 -0500
>>>>>> 
>>>>>> Changes made switch_cpp.cpp starting at Line 1000
>>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling
>>>>>> CoreSession::destroy\n");
>>>>>> if (session) {
>>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have
>>>>>> valid session\n");
>>>>>> 
>>>>>> if (!channel) {
>>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
>>>>>> undefined! Trying to get it!\n");
>>>>>> 
>>>>>> channel = switch_core_session_get_channel(session);
>>>>>> }
>>>>>> 
>>>>>> if (channel) {
>>>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>>>> "%s destroy/unlink session from object\n",
>>>>>> switch_channel_get_name(channel));
>>>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>>>>> "%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
>>>>>> switch_channel_set_private(channel, "CoreSession", NULL);
>>>>>> if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) &&
>>>>>> !switch_channel_test_flag(channel, CF_TRANSFER)) {
>>>>>> switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
>>>>>> }
>>>>>> } else {
>>>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
>>>>>> undefined! We Failed to get it!\n");
>>>>>> }
>>>>>> 
>>>>>> [CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
>>>>>> [CRIT] switch_cpp.cpp:1002 We still have valid session
>>>>>> [DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink
>>>>>> session from object
>>>>>> [DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9
>>>>>> destroy/unlink uuid
>>>>>> [DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX
>>>>>> Standard REPORTING, cause: NORMAL_CLEARING
>>>>>> [DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX )
>>>>>> State REPORTING going to sleep
>>>>>> [DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX )
>>>>>> State Change CS_REPORTING -> CS_DESTROY
>>>>>> [DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX
>>>>>> [BREAK]
>>>>>> [DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX
>>>>>> ) Locked, Waiting on external entities
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> channel remains hung
>>>>>> 
>>>>>> P
>>>>>> 
>>>>>> On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:
>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 chris at ghosttelecom.com  Wed Sep  5 17:54:34 2012
From: chris at ghosttelecom.com (Chris Martineau)
Date: Wed, 5 Sep 2012 13:54:34 +0000
Subject: [Freeswitch-users] SRTP performance overhead?
Message-ID: <F839D66EC0BCB04CB246EA21B8B4AF6B5BB758@DB3PRD0710MB369.eurprd07.prod.outlook.com>

Hi,

Does anyone have any information regarding the impact on freeswitch performance when using srtp?

i.e. if I can currently run a maximum of say 400 concurrent calls would this drop to 200 or is there no appreciable impact?

Many thanks

Chris
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From vetali100 at gmail.com  Wed Sep  5 19:51:01 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Wed, 5 Sep 2012 08:51:01 -0700
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
In-Reply-To: <CAFiqYum8tgGJHPdPgczeTb=S1zaVj9hyLXZCG1CCG0PLz-gdcw@mail.gmail.com>
References: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
	<CAFiqYum8tgGJHPdPgczeTb=S1zaVj9hyLXZCG1CCG0PLz-gdcw@mail.gmail.com>
Message-ID: <CACc3D0UnTprOEuqhF4_=VcOzPGCiLSsMf536AN+RKY+N-Q_JbQ@mail.gmail.com>

You can experience at least the following hangup causes from
FreeSWITCH  itself:
*ALLOTTED_TIMEOUT* - when you end the call by timer
*MEDIA_TIMEOUT* - when there is no RTP for some period (I would assume it
can happen if server lost network connection for some time)
*ATTENDED_TRANSFER* - not sure when it will happen, but I saw this in CDR
for an answered call

Case with media timeout is a very special one. In this case user might have
talked only for 60 seconds, but timeout happened after 1800 seconds (or how
is configured). You have no way to know how much time exactly the clients
have talked, because in the CDR you will see 1800 seconds. In this case I
think it will be fair to bill just for 1 minute or so and accept your
expenses.

I never saw other hangup cases for answered calls from FreeSWITCH itself.

Regards,
Vitalie




2012/9/5 Steven Ayre <steveayre at gmail.com>

> Do you have SIP traces for the call? If you're bridging then it's
> possible for the remote end to hangup the answered call with an ISDN
> clearing cause other than NORMAL_* (just like they can in ISDN
> signalling). For billing purposes I suggest you bill anything where
> billsec>0 rather than only NORMAL_CLEARING calls.
>
> -Steve
>
>
>
> On 4 September 2012 19:23, Muhammad Naseer Bhatti <nbhatti at gmail.com>
> wrote:
> > Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
> > NORMAL_CLEARING while the calls still have a valid duration and answered
> > time. For a typical billing scenario, calls are billed if (answered time)
> > billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the
> enumeration
> > would be other than normal and while the calls would be still a valid
> > answered call?
> >
> > 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 khuenm at vega.com.vn  Wed Sep  5 19:53:36 2012
From: khuenm at vega.com.vn (Khue Nguyen Minh)
Date: Wed, 5 Sep 2012 22:53:36 +0700
Subject: [Freeswitch-users] No Dialplan on answer channel
In-Reply-To: <CAKzWOxV1VUz2xGe50An9oWT53OoY0pkx_0Nf4QVLCeV-aSt1yw@mail.gmail.com>
References: <CAO-yT6hxe8eCcuT36WEjqmLDKYYfyGHnv3c2O_Opz3q6UMOhmA@mail.gmail.com>
	<CAKzWOxXyYZH8UUSDEVzvcF+avzFLYQaT-6sNbvsozSiW-FmsJQ@mail.gmail.com>
	<CALNZuwnKiXKR6S4eqiuOizu=1C+rBuGpxPnrzHDnU3cu9RvB=w@mail.gmail.com>
	<CAO-yT6gO4ymWJsM14H6EmSV0Vdc1AVJueTvABN0MOXkfe72gjw@mail.gmail.com>
	<CAKzWOxV1VUz2xGe50An9oWT53OoY0pkx_0Nf4QVLCeV-aSt1yw@mail.gmail.com>
Message-ID: <CAO-yT6h9JNBPDc2gMrLV-O-vxKmiefUVhctRRFu_GFJnBPOQEQ@mail.gmail.com>

Hi Michael,

This error has been resolved. Error is in my configuration file and rule
route in my sip proxy.

Thanks.

2012/9/4 Michael Collins <msc at freeswitch.org>

> "It doesn't work" is a big vague. Could you be more specific?
> -MC
>
>
> On Mon, Sep 3, 2012 at 6:49 PM, Khue Nguyen Minh <khuenm at vega.com.vn>wrote:
>
>> Hi Avi,
>>
>> My javascript work normal (I tested it in a xml dialplan). I still try
>> with originate sofia/external/1000 at somewhere.com &javascript('hello.js
>> /opt/freeswitch/sounds/2.mp3') but It doesn't work.
>>
>> Brs,
>> Khue Nguyen.
>>
>> 2012/8/31 Avi Marcus <avi at avimarcus.net>
>>
>>> 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
>>>  McAfee SiteAdvisor Warning
>>> This e-mail message contains potentially unsafe links to these sites:
>>> ostag.org [image: more info...]<https://www.siteadvisor.com/sites/http://www.ostag.org/-?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.229&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>>> freeswitchsolutions.com [image: more info...]<https://www.siteadvisor.com/sites/http://www.freeswitchsolutions.com/-?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.229&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>>>
>>>
>>> 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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Wed Sep  5 20:04:17 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Sep 2012 11:04:17 -0500
Subject: [Freeswitch-users] effective_callee_id_name behaviour
In-Reply-To: <CAAtWKeEXaWLfXL89QcwxHCHXWE07up13mgOwUUCytRNpv9EogQ@mail.gmail.com>
References: <CAAtWKeEXaWLfXL89QcwxHCHXWE07up13mgOwUUCytRNpv9EogQ@mail.gmail.com>
Message-ID: <CAKbxfG_=OZHu7hp9K9MJDobRFCce4Dse4n1BmVubb47tYpFncw@mail.gmail.com>

{origination_callee_id_name='test user',origination_callee_id_number=5551212}


On Wed, Sep 5, 2012 at 5:56 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Hello,
>
>   Our FreeSwitch is connected to a Nortel PBX via E1-SIP gateway
> (AudioCodes). This gateway supports the P-Asserted-ID field.
>
> With a vanilla configuration the following happens when a FS user calls a
> user on the Nortel:
> - During the ringing phase the name of the callee is "outbound call"
> - When the other side answers the name of the callee is set to the one sent
> from the Nortel.
>
> In order to have the name also during the ringing phase, I set calle_id_name
> and effective_callee_id_name.
> After doing so, the name is not changed after the remote user answers. I
> need it to change to the name sent
> from the Nortel, as it shows the name of the one who actually answered the
> phone.
>
> I also tried adding ignore_display_updates=false, but it didn't change the
> behaviour.
>
> Any idea how to do it?
>
>                         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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Sep  5 20:07:24 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Wed, 05 Sep 2012 09:07:24 -0700
Subject: [Freeswitch-users] Segfault from make current
In-Reply-To: <CAMWiMy9_wL0pFj79EPpODRwF7F2OxrHgowkZjvqYSV2b049SLg@mail.gmail.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
	<F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
	<CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>
	<95BF6B32-E362-4882-BA1C-F228058B4442@me.com>
	<CAKbxfG9cLhx-eTRx28WfVf-XEKcey+LGeEGsJbsqfFFoE4xqKw@mail.gmail.com>
	<7CD171F0-6BF1-4203-8FBF-93B6FE326E34@me.com>
	<CAMWiMy9_wL0pFj79EPpODRwF7F2OxrHgowkZjvqYSV2b049SLg@mail.gmail.com>
Message-ID: <65DC3856-D4F4-447E-BB29-37AC42CC466F@me.com>

Created http://jira.freeswitch.org/browse/FS-4594

On Sep 4, 2012, at 9:13 PM, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> Please report this as a bug on JIRA.
> 
> On Tue, Sep 4, 2012 at 8:50 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> Branching out from the libtiff string core dump info posted to http://pastebin.freeswitch.org/19839
>> 
>> 
>> On Sep 4, 2012, at 4:18 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> for that one see
>>> 
>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace
>>> 
>>> 
>>> On Tue, Sep 4, 2012 at 6:15 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> thanks that seem to get me past that point now I have another issue I will deal with later and start a new topic if I can not resolve seeing it's no longer related
>>>> 
>>>> kernel: [4076389.400073] freeswitch[13231]: segfault at 0 ip 00007f328d1cefa6 sp 00007fff2464d5d0 error 4 in libfreeswitch.so.1.0.0[7f328d13a000+210000]
>>>> 
>>>> 
>>>> On Sep 4, 2012, at 4:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>> 
>>>>> make spandsp-reconf
>>>>> 
>>>>> if that doesn't work you will need to manually fix libtiff
>>>>> 
>>>>> cd libs/tiff-4.0.2
>>>>> autoreconf -fi
>>>>> sh configure.gnu
>>>>> make
>>>>> cd ../..
>>>>> 
>>>>> resume as usual
>>>>> 
>>>>> 
>>>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Wed Sep  5 20:07:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Sep 2012 09:07:55 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxV+K8hf_DHwZvPYiFVAfChpgfAH2SaeWnTzn2FFGBqyiw@mail.gmail.com>

Hello all!

Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_05

We have a number of things to discuss as a community. Look forward to
seeing you there.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mike.burlingame at me.com  Wed Sep  5 20:14:20 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Wed, 05 Sep 2012 09:14:20 -0700
Subject: [Freeswitch-users] Segfault from make current
In-Reply-To: <65DC3856-D4F4-447E-BB29-37AC42CC466F@me.com>
References: <CAPNTnnrvLer_YDA0ft6b-9B5v0fEjjmaJFGd9pv4iBC_b58qZQ@mail.gmail.com>
	<CAMWiMy9RS5rQ_9JkHYDmxw8z5eqPwTUYo+Am_g+PoU3nJXKBnQ@mail.gmail.com>
	<CAPNTnno2UzBFWL_2RAXd1qG3HvEWUodh0=K67TFNzuaW0h9WeQ@mail.gmail.com>
	<1346728436032-7582519.post@n2.nabble.com>
	<1346730034461-7582520.post@n2.nabble.com>
	<CAPNTnnoua7Zkb-Xphxhq_QhRb+fBi4=e3nry=s1uJ0T4PZRUyg@mail.gmail.com>
	<F21D6309-4FFD-43C4-98C9-5DEA43903494@me.com>
	<CAKbxfG_Renv373oSCZjmD7ZKpMf2cLeYtACXE-ZV3PgcVbRASQ@mail.gmail.com>
	<95BF6B32-E362-4882-BA1C-F228058B4442@me.com>
	<CAKbxfG9cLhx-eTRx28WfVf-XEKcey+LGeEGsJbsqfFFoE4xqKw@mail.gmail.com>
	<7CD171F0-6BF1-4203-8FBF-93B6FE326E34@me.com>
	<CAMWiMy9_wL0pFj79EPpODRwF7F2OxrHgowkZjvqYSV2b049SLg@mail.gmail.com>
	<65DC3856-D4F4-447E-BB29-37AC42CC466F@me.com>
Message-ID: <7BC73152-914C-4024-9E1D-0C4A3B818F99@me.com>

This seems to have been resolved with this mornings make current - I am no longer getting the segfault 


On Sep 5, 2012, at 9:07 AM, Mike Burlingame <mike.burlingame at me.com> wrote:

> Created http://jira.freeswitch.org/browse/FS-4594
> 
> On Sep 4, 2012, at 9:13 PM, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
> 
>> Please report this as a bug on JIRA.
>> 
>> On Tue, Sep 4, 2012 at 8:50 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> Branching out from the libtiff string core dump info posted to http://pastebin.freeswitch.org/19839
>>> 
>>> 
>>> On Sep 4, 2012, at 4:18 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> 
>>>> for that one see
>>>> 
>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace
>>>> 
>>>> 
>>>> On Tue, Sep 4, 2012 at 6:15 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>> thanks that seem to get me past that point now I have another issue I will deal with later and start a new topic if I can not resolve seeing it's no longer related
>>>>> 
>>>>> kernel: [4076389.400073] freeswitch[13231]: segfault at 0 ip 00007f328d1cefa6 sp 00007fff2464d5d0 error 4 in libfreeswitch.so.1.0.0[7f328d13a000+210000]
>>>>> 
>>>>> 
>>>>> On Sep 4, 2012, at 4:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>>> 
>>>>>> make spandsp-reconf
>>>>>> 
>>>>>> if that doesn't work you will need to manually fix libtiff
>>>>>> 
>>>>>> cd libs/tiff-4.0.2
>>>>>> autoreconf -fi
>>>>>> sh configure.gnu
>>>>>> make
>>>>>> cd ../..
>>>>>> 
>>>>>> resume as usual
>>>>>> 
>>>>>> 
>>>>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep  5 20:50:36 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 5 Sep 2012 12:50:36 -0400
Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file
In-Reply-To: <50472735.7080601@gmail.com>
References: <5046EDC5.8080605@gmail.com>
	<CAFiqYuk-g+xxQ2g8cGo08q2_mgHvj8hkD_NPpCy58zHHZqKEiQ@mail.gmail.com>
	<50472735.7080601@gmail.com>
Message-ID: <1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com>

If freeswitch is not already using one of its directory defines to pass down to the library for where this file should be, someone please open a jira issue to correct this.  This should be using one of dirs we define out of configure appropriately so it has the right permissions.

Thanks
Mike

On Sep 5, 2012, at 6:19 AM, Vbvbrj <vbvbrj at gmail.com> wrote:

> On 05.09.2012 11:39, Steven Ayre wrote:
>> It's a log message from within the libzrtp library.
>> 
>> It means it cannot open the cache file. The log message contains the
>> filename it tried to open, but it looks like you've removed the end of
>> the lines with that information.
> 
> No, I didn't remove any symbol from the log output. Realy it does not 
> show the file name.
> 
>> The default filename within the library is
>> "./zrtp_def_cache_path.dat". That'd be within the cwd FS is running
>> with, which you can find with "ls -ld /proc/PID/cwd" where PID is the
>> PID freeswitch is running as.
> 
> Yes, the cmd of FS is running is correct and the username under which FS 
> is running does not have write permission except some subfolders, for 
> ex. log sobfolder. Then how to specify the zrtp cache file path and name?
> 
> -- 



From mike.burlingame at me.com  Wed Sep  5 21:18:39 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Wed, 05 Sep 2012 10:18:39 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <071E375F-B14D-46DA-899F-38F3725F0C92@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
Message-ID: <1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>

as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg

I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help


Call Flow with out {sip_wait_for_aleg_ack=true}
  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
  0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
  0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
  0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
  0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
  0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
  4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description
  4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK

Call Flow with {sip_wait_for_aleg_ack=true} enabled
  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
  0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
  0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
  0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
  0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
  0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description
 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK

On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:

> No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
> 
> 
> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> 
>> 
>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> Cool I will nail that up on my test box and see if that works
>> 
>> Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
>> 
>> -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

-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Sep  5 22:10:57 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Sep 2012 13:10:57 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <1D1B9033-340F-469C-8A79-D33A79071BFF@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
Message-ID: <CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>

update and try again, if it still doesn't work open a jira


On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> as promised here is the update testing and enabling
> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>
> I can open a JIRA case on this and provide the console log file / PCAP's ect
> if that would help
>
>
> Call Flow with out {sip_wait_for_aleg_ack=true}
>   0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> sip:+13605551212 at A-LEG:5060, with session description
>   0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>   0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> sip:13605551212 at B-Leg, with session description
>   0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>   0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>   0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>   4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>   4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> description
>   4.533691 A-LEG -> FreeSwitch SIP Request: ACK
> sip:+13605551212 at FreeSwitch:5070;transport=udp
>  19.953061 A-LEG -> FreeSwitch SIP Request: BYE
> sip:+13605551212 at FreeSwitch:5070;transport=udp
>  19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>  19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>  19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>
> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>   0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> sip:+13605551212 at A-LEG:5060, with session description
>   0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>   0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> sip:13605551212 at B-Leg, with session description
>   0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>   0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>   0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>   4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>   5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>   5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>   6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>   7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>  11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
>  15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>  15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> description
>  15.173265 A-LEG -> FreeSwitch SIP Request: ACK
> sip:+13605551212 at FreeSwitch:5070;transport=udp
>  29.566504 A-LEG -> FreeSwitch SIP Request: BYE
> sip:+13605551212 at FreeSwitch:5070;transport=udp
>  29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>  29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>  29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>
> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>
> No worries I will be out this weekend for the long weekend I will work on
> getting the test box upgraded and a test case setup on Tuesday I will report
> back the results mid to late next week and provided everything works as I
> hope it will I will be happy to pay the Wiki tax :)
>
>
> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
>
> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
> wrote:
>>
>> Cool I will nail that up on my test box and see if that works
>>
> Please report back on whether it works or not and then be prepared to pay
> the wiki tax. :) I'll be glad to assist with getting this documented
> although I think you're in the best position to give that documentation some
> real-world context.
>
> -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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Wed Sep  5 22:18:31 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Sep 2012 13:18:31 -0500
Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file
In-Reply-To: <1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com>
References: <5046EDC5.8080605@gmail.com>
	<CAFiqYuk-g+xxQ2g8cGo08q2_mgHvj8hkD_NPpCy58zHHZqKEiQ@mail.gmail.com>
	<50472735.7080601@gmail.com>
	<1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com>
Message-ID: <CAKbxfG9qcM7TE3KHnG60twJJvhU22uPnvnTzavaWynrK-KUN8w@mail.gmail.com>

switch_snprintf(zrtp_cache_path, sizeof(zrtp_cache_path),
"%s%szrtp.dat", SWITCH_GLOBAL_dirs.db_dir, SWITCH_PATH_SEPARATOR);

it should go in /usr/local/freeswitch/db

does the user running the process have ownership of /usr/local/freeswitch ?

On Wed, Sep 5, 2012 at 11:50 AM, Michael Jerris <mike at jerris.com> wrote:
> If freeswitch is not already using one of its directory defines to pass down to the library for where this file should be, someone please open a jira issue to correct this.  This should be using one of dirs we define out of configure appropriately so it has the right permissions.
>
> Thanks
> Mike
>
> On Sep 5, 2012, at 6:19 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
>
>> On 05.09.2012 11:39, Steven Ayre wrote:
>>> It's a log message from within the libzrtp library.
>>>
>>> It means it cannot open the cache file. The log message contains the
>>> filename it tried to open, but it looks like you've removed the end of
>>> the lines with that information.
>>
>> No, I didn't remove any symbol from the log output. Realy it does not
>> show the file name.
>>
>>> The default filename within the library is
>>> "./zrtp_def_cache_path.dat". That'd be within the cwd FS is running
>>> with, which you can find with "ls -ld /proc/PID/cwd" where PID is the
>>> PID freeswitch is running as.
>>
>> Yes, the cmd of FS is running is correct and the username under which FS
>> is running does not have write permission except some subfolders, for
>> ex. log sobfolder. Then how to specify the zrtp cache file path and name?
>>
>> --
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Sep  5 22:37:33 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Wed, 05 Sep 2012 11:37:33 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
Message-ID: <283776BC-0455-4C16-8904-13396619CA3A@me.com>

The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?

commit d45db898339e1b2212f5befff1af714abcec034f
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Sep 5 13:11:32 2012 -0500

  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
  0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
  0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
  0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
  0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
  0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist


On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> update and try again, if it still doesn't work open a jira
> 
> 
> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> as promised here is the update testing and enabling
>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>> 
>> I can open a JIRA case on this and provide the console log file / PCAP's ect
>> if that would help
>> 
>> 
>> Call Flow with out {sip_wait_for_aleg_ack=true}
>>  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>> sip:+13605551212 at A-LEG:5060, with session description
>>  0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>  0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>> sip:13605551212 at B-Leg, with session description
>>  0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>  0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>  0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>  0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>  4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>>  4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>  4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>> description
>>  4.533691 A-LEG -> FreeSwitch SIP Request: ACK
>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>> 
>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>>  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>> sip:+13605551212 at A-LEG:5060, with session description
>>  0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>  0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>> sip:13605551212 at B-Leg, with session description
>>  0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>  0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>  0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>  0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>  4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>>  4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>>  5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>>  5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>>  6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>>  7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> description
>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>> description
>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>> 
>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> 
>> No worries I will be out this weekend for the long weekend I will work on
>> getting the test box upgraded and a test case setup on Tuesday I will report
>> back the results mid to late next week and provided everything works as I
>> hope it will I will be happy to pay the Wiki tax :)
>> 
>> 
>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>> 
>> 
>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
>> wrote:
>>> 
>>> Cool I will nail that up on my test box and see if that works
>>> 
>> Please report back on whether it works or not and then be prepared to pay
>> the wiki tax. :) I'll be glad to assist with getting this documented
>> although I think you're in the best position to give that documentation some
>> real-world context.
>> 
>> -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
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 vbvbrj at gmail.com  Wed Sep  5 22:54:37 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 05 Sep 2012 21:54:37 +0300
Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file
In-Reply-To: <CAKbxfG9qcM7TE3KHnG60twJJvhU22uPnvnTzavaWynrK-KUN8w@mail.gmail.com>
References: <5046EDC5.8080605@gmail.com>
	<CAFiqYuk-g+xxQ2g8cGo08q2_mgHvj8hkD_NPpCy58zHHZqKEiQ@mail.gmail.com>
	<50472735.7080601@gmail.com>
	<1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com>
	<CAKbxfG9qcM7TE3KHnG60twJJvhU22uPnvnTzavaWynrK-KUN8w@mail.gmail.com>
Message-ID: <50479FED.9060705@gmail.com>

On 05.09.2012 21:18, Anthony Minessale wrote:
> switch_snprintf(zrtp_cache_path, sizeof(zrtp_cache_path),
> "%s%szrtp.dat", SWITCH_GLOBAL_dirs.db_dir, SWITCH_PATH_SEPARATOR);
>
> it should go in /usr/local/freeswitch/db
>
> does the user running the process have ownership of /usr/local/freeswitch ?

The user "freeswitch" and which FS is running does not have the 
ownership, but the group it belongs has rx access and rws access to the 
log,db,recording and some other subdirectories. FS was build and 
isntalled with --prefix.

-- 
Mimiko desu.


From vbvbrj at gmail.com  Wed Sep  5 22:55:40 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 05 Sep 2012 21:55:40 +0300
Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file
In-Reply-To: <CAKbxfG9qcM7TE3KHnG60twJJvhU22uPnvnTzavaWynrK-KUN8w@mail.gmail.com>
References: <5046EDC5.8080605@gmail.com>
	<CAFiqYuk-g+xxQ2g8cGo08q2_mgHvj8hkD_NPpCy58zHHZqKEiQ@mail.gmail.com>
	<50472735.7080601@gmail.com>
	<1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com>
	<CAKbxfG9qcM7TE3KHnG60twJJvhU22uPnvnTzavaWynrK-KUN8w@mail.gmail.com>
Message-ID: <5047A02C.5090007@gmail.com>

Oh, FS user can write to /opt/freeswitch/bd.

-- 
Mimiko desu.


From anthony.minessale at gmail.com  Wed Sep  5 23:01:21 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Sep 2012 14:01:21 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <283776BC-0455-4C16-8904-13396619CA3A@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
Message-ID: <CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>

ok,

update one more time, if it still does not work just go right to jira
with the latest (not before today's changes)


On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
>
> commit d45db898339e1b2212f5befff1af714abcec034f
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Wed Sep 5 13:11:32 2012 -0500
>
>   0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>   0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
>   0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>   0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>   0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>   0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>   5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>   5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>   6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>   6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>   8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
>  24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>  24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>  24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>  24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>  27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>  27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
>  27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>  27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>  27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>  27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>  27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>  31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
>  49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
>  49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
>  49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>  49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>  49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>  49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>
>
> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> update and try again, if it still doesn't work open a jira
>>
>>
>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> as promised here is the update testing and enabling
>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
>>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>>>
>>> I can open a JIRA case on this and provide the console log file / PCAP's ect
>>> if that would help
>>>
>>>
>>> Call Flow with out {sip_wait_for_aleg_ack=true}
>>>  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>> sip:+13605551212 at A-LEG:5060, with session description
>>>  0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>  0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>> sip:13605551212 at B-Leg, with session description
>>>  0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>  0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>  0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>  0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>  4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>>  4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>  4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>> description
>>>  4.533691 A-LEG -> FreeSwitch SIP Request: ACK
>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>
>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>>>  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>> sip:+13605551212 at A-LEG:5060, with session description
>>>  0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>  0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>> sip:13605551212 at B-Leg, with session description
>>>  0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>  0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>  0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>  0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>  4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>>  4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>>  5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>>  5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>>  6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>>  7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> description
>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>> description
>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>
>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>
>>> No worries I will be out this weekend for the long weekend I will work on
>>> getting the test box upgraded and a test case setup on Tuesday I will report
>>> back the results mid to late next week and provided everything works as I
>>> hope it will I will be happy to pay the Wiki tax :)
>>>
>>>
>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>
>>>
>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
>>> wrote:
>>>>
>>>> Cool I will nail that up on my test box and see if that works
>>>>
>>> Please report back on whether it works or not and then be prepared to pay
>>> the wiki tax. :) I'll be glad to assist with getting this documented
>>> although I think you're in the best position to give that documentation some
>>> real-world context.
>>>
>>> -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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 mike.burlingame at me.com  Wed Sep  5 23:28:42 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Wed, 05 Sep 2012 12:28:42 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
Message-ID: <927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>

Looks much much better Thank you -- Now to conduct more testing 

 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-Leg:5060, with session description
  0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying
  0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
  0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
  0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
  0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
  5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
  5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description
  5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
  5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK
 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK



On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> ok,
> 
> update one more time, if it still does not work just go right to jira
> with the latest (not before today's changes)
> 
> 
> On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
>> 
>> commit d45db898339e1b2212f5befff1af714abcec034f
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date:   Wed Sep 5 13:11:32 2012 -0500
>> 
>>  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>>  0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>  0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>  0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>  0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>  0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>  0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>  4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>  5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>  5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>  6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>  6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>  8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
>> 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>> 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>> 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
>> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>> 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
>> 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
>> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
>> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>> 
>> 
>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> update and try again, if it still doesn't work open a jira
>>> 
>>> 
>>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> as promised here is the update testing and enabling
>>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
>>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
>>>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>>>> 
>>>> I can open a JIRA case on this and provide the console log file / PCAP's ect
>>>> if that would help
>>>> 
>>>> 
>>>> Call Flow with out {sip_wait_for_aleg_ack=true}
>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>>> sip:+13605551212 at A-LEG:5060, with session description
>>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>>> sip:13605551212 at B-Leg, with session description
>>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>> 
>>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>>> sip:+13605551212 at A-LEG:5060, with session description
>>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>>> sip:13605551212 at B-Leg, with session description
>>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>>> description
>>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>> 
>>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> 
>>>> No worries I will be out this weekend for the long weekend I will work on
>>>> getting the test box upgraded and a test case setup on Tuesday I will report
>>>> back the results mid to late next week and provided everything works as I
>>>> hope it will I will be happy to pay the Wiki tax :)
>>>> 
>>>> 
>>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>> 
>>>> 
>>>> 
>>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
>>>> wrote:
>>>>> 
>>>>> Cool I will nail that up on my test box and see if that works
>>>>> 
>>>> Please report back on whether it works or not and then be prepared to pay
>>>> the wiki tax. :) I'll be glad to assist with getting this documented
>>>> although I think you're in the best position to give that documentation some
>>>> real-world context.
>>>> 
>>>> -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
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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



From sos at sokhapkin.dyndns.org  Thu Sep  6 02:32:03 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 05 Sep 2012 18:32:03 -0400
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
Message-ID: <1541775.1ZdC32xc96@sos>

 Anthony, shouldn't it be the default behavior? I see no reason to send ACK to 
B leg before we get ACK from A-leg.

On Wednesday 05 September 2012 12:28:42 Mike Burlingame wrote:
> Looks much much better Thank you -- Now to conduct more testing
> 
>  0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE
> sip:+13605551212 at A-Leg:5060, with session description 0.000639 FreeSwitch
> -> A-Leg SIP Status: 100 Trying
>   0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> sip:13605551212 at B-Leg, with session description 0.051351 B-Leg ->
> FreeSwitch SIP Status: 100 Giving a try
>   0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
>   0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>   5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with
> session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK
> sip:+13605551212 at FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg
> SIP Request: ACK sip:13605551212 at B-Leg 12.027026 A-Leg -> FreeSwitch SIP
> Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp 12.029232
> FreeSwitch -> A-Leg SIP Status: 200 OK
>  12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>  12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
> 
> On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale at gmail.com> 
wrote:
> > ok,
> > 
> > update one more time, if it still does not work just go right to jira
> > with the latest (not before today's changes)
> > 
> > On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> 
wrote:
> >> The change seems to have broken the ability for the call to connect -
> >> would you like me to open a jira up with the current log files or before
> >> the change was made today?
> >> 
> >> commit d45db898339e1b2212f5befff1af714abcec034f
> >> Author: Anthony Minessale <anthm at freeswitch.org>
> >> Date:   Wed Sep 5 13:11:32 2012 -0500
> >> 
> >>  0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> >>  sip:+13605551212 at A-LEG:5060, with session description 0.002715
> >>  FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>  0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >>  sip:13605551212 at B-Leg, with session description 0.062976 B-Leg ->
> >>  FreeSwitch SIP Status: 100 Giving a try
> >>  0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>  0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>  0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>  4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>  description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with
> >>  session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200
> >>  OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP
> >>  Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch
> >>  SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg ->
> >>  FreeSwitch SIP/SDP Status: 200 OK, with session description>> 
> >> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >> description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily
> >> Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK
> >> sip:+13605551212 at A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request:
> >> ACK sip:13605551212 at B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE
> >> sip:13605551212 at B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call
> >> leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP
> >> Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
> >> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >> sip:13605551212 at B-Leg, with session description 27.123445 B-Leg ->
> >> FreeSwitch SIP Status: 100 Giving a try
> >> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >> description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with
> >> session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200
> >> OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP
> >> Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch
> >> SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg ->
> >> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852
> >> B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL
> >> sip:+13605551212 at A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status:
> >> 200 OK
> >> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
> >> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK
> >> sip:+13605551212 at A-LEG:5060
> >> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does
> >> not exist>> 
> >> On Sep 5, 2012, at 11:10 AM, Anthony Minessale 
<anthony.minessale at gmail.com> wrote:
> >>> update and try again, if it still doesn't work open a jira
> >>> 
> >>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame 
<mike.burlingame at me.com> wrote:
> >>>> as promised here is the update testing and enabling
> >>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts
> >>>> to
> >>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from
> >>>> the
> >>>> B-Leg as well as an abnormally long delay in getting an ACK from the
> >>>> A-Leg
> >>>> 
> >>>> I can open a JIRA case on this and provide the console log file /
> >>>> PCAP's ect if that would help
> >>>> 
> >>>> 
> >>>> Call Flow with out {sip_wait_for_aleg_ack=true}
> >>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> >>>> sip:+13605551212 at A-LEG:5060, with session description
> >>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >>>> sip:13605551212 at B-Leg, with session description
> >>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
> >>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
> >>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>>> 
> >>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
> >>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> >>>> sip:+13605551212 at A-LEG:5060, with session description
> >>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >>>> sip:13605551212 at B-Leg, with session description
> >>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> >>>> description
> >>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
> >>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
> >>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>>> 
> >>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com>
> >>>> wrote:
> >>>> 
> >>>> No worries I will be out this weekend for the long weekend I will work
> >>>> on
> >>>> getting the test box upgraded and a test case setup on Tuesday I will
> >>>> report back the results mid to late next week and provided everything
> >>>> works as I hope it will I will be happy to pay the Wiki tax :)
> >>>> 
> >>>> 
> >>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org>
> >>>> wrote:
> >>>> 
> >>>> 
> >>>> 
> >>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame
> >>>> <mike.burlingame at me.com>
> >>>> 
> >>>> wrote:
> >>>>> Cool I will nail that up on my test box and see if that works
> >>>> 
> >>>> Please report back on whether it works or not and then be prepared to
> >>>> pay
> >>>> the wiki tax. :) I'll be glad to assist with getting this documented
> >>>> although I think you're in the best position to give that documentation
> >>>> some real-world context.
> >>>> 
> >>>> -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-user
> >>>> s
> >>>> http://www.freeswitch.org
> >>>> 
> >>>> 
> >>>> _______________________________________________________________________
> >>>> __
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>> 
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>> s
> >>>> http://www.freeswitch.org
> >>>> 
> >>>> 
> >>>> 
> >>>> _______________________________________________________________________
> >>>> __
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>> 
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>> s
> >>>> 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


From kris at kriskinc.com  Thu Sep  6 02:34:38 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 5 Sep 2012 18:34:38 -0400
Subject: [Freeswitch-users] Bat-Phone or Moscow RED PHONE Hotline
In-Reply-To: <CAMwZ-tLHNSGuz07y8jGA1ZmjUda7CfifF+_tQPEOoUNVM=z22Q@mail.gmail.com>
References: <006401cd8ad7$28739e80$795adb80$@bizfocused.com>
	<CAMwZ-t+66=n992kX4sXwVYE6dx25iCqgLTwBLx27xgz+LMoWkw@mail.gmail.com>
	<CAKzWOxVPgwn3kObWsgS22svKqQOWnfwvWC9y_XyYRLczPyeDKg@mail.gmail.com>
	<CAKDfjgc6DzwAk7LN=20nVQSeXXLgqAO3U6E15Z0bKSf+TLD3HA@mail.gmail.com>
	<CAMwZ-tLHNSGuz07y8jGA1ZmjUda7CfifF+_tQPEOoUNVM=z22Q@mail.gmail.com>
Message-ID: <CAKDfjgcLm+eTJDxQcVeFrCJjf6jhKcdDW=o05ZFAfwbnAEg=5Q@mail.gmail.com>

It's a little old at this point and that killhofner (or whatever) guy
is a bit of a clown but everyone else did a pretty good job ;).

On Tue, Sep 4, 2012 at 10:41 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Tue, Sep 4, 2012 at 3:29 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>> Sipura dialplans (especially) are remarkably powerful.  Here is a
>> Google Books link to a few examples:
>
>
> Heh, I have that book, but I've never picked it up; maybe I should.
>
> Can you vouch for any of the authors? ;)
>
>
> Gabe
>

-- 
Kristian Kielhofner


From matt.putnam at lightspar.com  Thu Sep  6 02:21:58 2012
From: matt.putnam at lightspar.com (Matt Putnam)
Date: Wed, 5 Sep 2012 17:21:58 -0500
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
Message-ID: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>

This may have been asked before but my research hasn't turned up any results. I am currently testing freeswitch as a replacement to our current asterisk gateways. I think I have the basics of ip based trunks down but I am running into an issue with trunks registering to the freeswitch box. In asterisk it was a fairly simple process of giving the trunk a username and password and setting it to dynamic to get it to register. What would be the equivalent in freeswitch that would allow me to have customer trunks register to the system?


Thanks,
Matt Putnam

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From msc at freeswitch.org  Thu Sep  6 03:48:48 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Sep 2012 16:48:48 -0700
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
Message-ID: <CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>

Hi Matt,

Welcome to FreeSWITCH!

If you want something to register with FreeSWITCH then simply add an entry
in the user directory. Whether it's a "user" or a "trunk" really doesn't
matter - it's just a SIP registration. Look in
conf/directory/default/1000.xml (if using the example "vanilla"
configuration) for a taste of what you need. Let us know if you have any
other questions or join us in #freeswitch on irc.freenode.net.

-MC (IRC:mercutioviz)

On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com>wrote:

> This may have been asked before but my research hasn?t turned up any
> results. I am currently testing freeswitch as a replacement to our current
> asterisk gateways. I think I have the basics of ip based trunks down but I
> am running into an issue with trunks registering to the freeswitch box. In
> asterisk it was a fairly simple process of giving the trunk a username and
> password and setting it to dynamic to get it to register. What would be the
> equivalent in freeswitch that would allow me to have customer trunks
> register to the system? ****
>
> ** **
>
> ** **
>
> Thanks,****
>
> Matt Putnam****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nbhatti at gmail.com  Thu Sep  6 05:45:04 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Thu, 6 Sep 2012 04:45:04 +0300
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
In-Reply-To: <CACc3D0UnTprOEuqhF4_=VcOzPGCiLSsMf536AN+RKY+N-Q_JbQ@mail.gmail.com>
References: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
	<CAFiqYum8tgGJHPdPgczeTb=S1zaVj9hyLXZCG1CCG0PLz-gdcw@mail.gmail.com>
	<CACc3D0UnTprOEuqhF4_=VcOzPGCiLSsMf536AN+RKY+N-Q_JbQ@mail.gmail.com>
Message-ID: <CACak_ywyMXaRRjKQ-tvDjPWWK84ZVd3K+nqywypWt_saU_vijg@mail.gmail.com>

I don't have the SIP traces right now, but I see causes like MEDIA_TIMEOUT,
*DESTINATION_OUT_OF_ORDER* , *NORMAL_UNSPECIFIED* , *NETWORK_OUT_OF_ORDER*
and *RECOVERY_ON_TIMER_EXPIRE*  and the calls were still good. We are not
talking about a few, they were around 50K + calls for such reasons in 24
hours. I'll try to capture SIP for any new calls. Was just curious what's
going on.

-B

On Wed, Sep 5, 2012 at 6:51 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:

> You can experience at least the following hangup causes from
> FreeSWITCH  itself:
> *ALLOTTED_TIMEOUT* - when you end the call by timer
> *MEDIA_TIMEOUT* - when there is no RTP for some period (I would assume it
> can happen if server lost network connection for some time)
> *ATTENDED_TRANSFER* - not sure when it will happen, but I saw this in CDR
> for an answered call
>
> Case with media timeout is a very special one. In this case user might
> have talked only for 60 seconds, but timeout happened after 1800 seconds
> (or how is configured). You have no way to know how much time exactly the
> clients have talked, because in the CDR you will see 1800 seconds. In this
> case I think it will be fair to bill just for 1 minute or so and accept
> your expenses.
>
> I never saw other hangup cases for answered calls from FreeSWITCH itself.
>
> Regards,
> Vitalie
>
>
>
>
> 2012/9/5 Steven Ayre <steveayre at gmail.com>
>
>> Do you have SIP traces for the call? If you're bridging then it's
>> possible for the remote end to hangup the answered call with an ISDN
>> clearing cause other than NORMAL_* (just like they can in ISDN
>> signalling). For billing purposes I suggest you bill anything where
>> billsec>0 rather than only NORMAL_CLEARING calls.
>>
>> -Steve
>>
>>
>>
>> On 4 September 2012 19:23, Muhammad Naseer Bhatti <nbhatti at gmail.com>
>> wrote:
>> > Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
>> > NORMAL_CLEARING while the calls still have a valid duration and answered
>> > time. For a typical billing scenario, calls are billed if (answered
>> time)
>> > billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the
>> enumeration
>> > would be other than normal and while the calls would be still a valid
>> > answered call?
>> >
>> > 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 djbinter at gmail.com  Thu Sep  6 05:51:44 2012
From: djbinter at gmail.com (Dorn DJBinter)
Date: Wed, 5 Sep 2012 18:51:44 -0700
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
In-Reply-To: <CACak_ywyMXaRRjKQ-tvDjPWWK84ZVd3K+nqywypWt_saU_vijg@mail.gmail.com>
References: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
	<CAFiqYum8tgGJHPdPgczeTb=S1zaVj9hyLXZCG1CCG0PLz-gdcw@mail.gmail.com>
	<CACc3D0UnTprOEuqhF4_=VcOzPGCiLSsMf536AN+RKY+N-Q_JbQ@mail.gmail.com>
	<CACak_ywyMXaRRjKQ-tvDjPWWK84ZVd3K+nqywypWt_saU_vijg@mail.gmail.com>
Message-ID: <-6205500302962635460@unknownmsgid>

Do you have call failover when you bridge?  It could be causes from the 1st
gateway that failed, but your call continued to complete on the next one.

Sent from my iPad

On Sep 5, 2012, at 6:47 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>
wrote:

I don't have the SIP traces right now, but I see causes like MEDIA_TIMEOUT,
*DESTINATION_OUT_OF_ORDER* , *NORMAL_UNSPECIFIED* , *NETWORK_OUT_OF_ORDER*
and *RECOVERY_ON_TIMER_EXPIRE*  and the calls were still good. We are not
talking about a few, they were around 50K + calls for such reasons in 24
hours. I'll try to capture SIP for any new calls. Was just curious what's
going on.

-B

On Wed, Sep 5, 2012 at 6:51 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:

> You can experience at least the following hangup causes from
> FreeSWITCH  itself:
> *ALLOTTED_TIMEOUT* - when you end the call by timer
> *MEDIA_TIMEOUT* - when there is no RTP for some period (I would assume it
> can happen if server lost network connection for some time)
> *ATTENDED_TRANSFER* - not sure when it will happen, but I saw this in CDR
> for an answered call
>
> Case with media timeout is a very special one. In this case user might
> have talked only for 60 seconds, but timeout happened after 1800 seconds
> (or how is configured). You have no way to know how much time exactly the
> clients have talked, because in the CDR you will see 1800 seconds. In this
> case I think it will be fair to bill just for 1 minute or so and accept
> your expenses.
>
> I never saw other hangup cases for answered calls from FreeSWITCH itself.
>
> Regards,
> Vitalie
>
>
>
>
> 2012/9/5 Steven Ayre <steveayre at gmail.com>
>
>> Do you have SIP traces for the call? If you're bridging then it's
>> possible for the remote end to hangup the answered call with an ISDN
>> clearing cause other than NORMAL_* (just like they can in ISDN
>> signalling). For billing purposes I suggest you bill anything where
>> billsec>0 rather than only NORMAL_CLEARING calls.
>>
>> -Steve
>>
>>
>>
>> On 4 September 2012 19:23, Muhammad Naseer Bhatti <nbhatti at gmail.com>
>> wrote:
>> > Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
>> > NORMAL_CLEARING while the calls still have a valid duration and answered
>> > time. For a typical billing scenario, calls are billed if (answered
>> time)
>> > billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the
>> enumeration
>> > would be other than normal and while the calls would be still a valid
>> > answered call?
>> >
>> > 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 nbhatti at gmail.com  Thu Sep  6 06:03:34 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Thu, 6 Sep 2012 05:03:34 +0300
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
In-Reply-To: <-6205500302962635460@unknownmsgid>
References: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
	<CAFiqYum8tgGJHPdPgczeTb=S1zaVj9hyLXZCG1CCG0PLz-gdcw@mail.gmail.com>
	<CACc3D0UnTprOEuqhF4_=VcOzPGCiLSsMf536AN+RKY+N-Q_JbQ@mail.gmail.com>
	<CACak_ywyMXaRRjKQ-tvDjPWWK84ZVd3K+nqywypWt_saU_vijg@mail.gmail.com>
	<-6205500302962635460@unknownmsgid>
Message-ID: <CACak_yzbdE6ifgvdBPBDp78zPD7OHj5tPBzjT2XOiE1pLk5r3g@mail.gmail.com>

We do bridge multiple gateways and capture failed gateways but the hangup
cause shouldn't be the from the last tried gateway? That last gateway
should have a good call because continue_on_fail = false. If the call fails
1st gw, the hangup cause in the cdr should come from the last tried gw
rather than the 1st one.

-B

On Thu, Sep 6, 2012 at 4:51 AM, Dorn DJBinter <djbinter at gmail.com> wrote:

> Do you have call failover when you bridge?  It could be causes from the
> 1st gateway that failed, but your call continued to complete on the next
> one.
>
> Sent from my iPad
>
> On Sep 5, 2012, at 6:47 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>
> wrote:
>
> I don't have the SIP traces right now, but I see causes like
> MEDIA_TIMEOUT, *DESTINATION_OUT_OF_ORDER* , *NORMAL_UNSPECIFIED* , *
> NETWORK_OUT_OF_ORDER*  and *RECOVERY_ON_TIMER_EXPIRE*  and the calls were
> still good. We are not talking about a few, they were around 50K + calls
> for such reasons in 24 hours. I'll try to capture SIP for any new calls.
> Was just curious what's going on.
>
> -B
>
> On Wed, Sep 5, 2012 at 6:51 PM, Vitalie Colosov <vetali100 at gmail.com>wrote:
>
>> You can experience at least the following hangup causes from
>> FreeSWITCH  itself:
>> *ALLOTTED_TIMEOUT* - when you end the call by timer
>> *MEDIA_TIMEOUT* - when there is no RTP for some period (I would assume
>> it can happen if server lost network connection for some time)
>> *ATTENDED_TRANSFER* - not sure when it will happen, but I saw this in
>> CDR for an answered call
>>
>> Case with media timeout is a very special one. In this case user might
>> have talked only for 60 seconds, but timeout happened after 1800 seconds
>> (or how is configured). You have no way to know how much time exactly the
>> clients have talked, because in the CDR you will see 1800 seconds. In this
>> case I think it will be fair to bill just for 1 minute or so and accept
>> your expenses.
>>
>> I never saw other hangup cases for answered calls from FreeSWITCH itself.
>>
>> Regards,
>> Vitalie
>>
>>
>>
>>
>> 2012/9/5 Steven Ayre <steveayre at gmail.com>
>>
>>> Do you have SIP traces for the call? If you're bridging then it's
>>> possible for the remote end to hangup the answered call with an ISDN
>>> clearing cause other than NORMAL_* (just like they can in ISDN
>>> signalling). For billing purposes I suggest you bill anything where
>>> billsec>0 rather than only NORMAL_CLEARING calls.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 4 September 2012 19:23, Muhammad Naseer Bhatti <nbhatti at gmail.com>
>>> wrote:
>>> > Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
>>> > NORMAL_CLEARING while the calls still have a valid duration and
>>> answered
>>> > time. For a typical billing scenario, calls are billed if (answered
>>> time)
>>> > billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the
>>> enumeration
>>> > would be other than normal and while the calls would be still a valid
>>> > answered call?
>>> >
>>> > 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 lconroy at insensate.co.uk  Thu Sep  6 14:33:57 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Thu, 6 Sep 2012 11:33:57 +0100
Subject: [Freeswitch-users] 1.2 stable x64 installer build issue?
Message-ID: <72CE317B-F7FB-4E9D-A365-B8EDC80045B1@insensate.co.uk>

Hi there,
I've had issues with 1.2 stable installer build with VS 2010 pro; I'm unsure if this is a bug in the Setup.wixproj (installer build) file or an error in user.
Please bear with me -- I've just switched from using an old TiBook with Leopard as that required too many system changes to work => I have little idea what I'm doing.

Steps for 1.2 stable x64 build:
-	Git clone -b v1.2.stable, & fire up VS2010pro+Wix3.5 [note -- not express, and without the SDK7.1]
-	Select x64 Release; Unload project mod_cepstral, and delete cepstral from Setup references
-	build -> works fine

-	select x64 Setup -> fails

-	change one of Setup's pre-build event commands from:
"$(WIX)bin\heat.exe" dir "$(ProjectDir)..\..\Win32\Release\sounds" -cg FreeSWITCHSoundFiles8 -gg -scom -sreg -sfrag -srd -dr SOUNDLOCATION -var var.FreeSWITCHSoundFilesDir -out "$(ProjectDir)Fragments\FreeSWITCHSoundFiles8.wxs"
to:
"$(WIX)bin\heat.exe" dir "$(ProjectDir)..\..\x64\Release\sounds" -cg FreeSWITCHSoundFiles8 -gg -scom -sreg -sfrag -srd -dr SOUNDLOCATION -var var.FreeSWITCHSoundFilesDir -out "$(ProjectDir)Fragments\FreeSWITCHSoundFiles8.wxs"

-	build -> works fine

=> Making this pre-build change in VS2010 seems to have worked for me.
[unsurprisingly, with an x64 build, the sounds are placed into the x64/Release path]

I *think* that this pre-build command is generated from Setup.wixproj (near the end of that file).
-	Git clone -b v1.2.stable;
-	edit freeswitch\w32\Setup\Setup.wixproj to change Win32\Release\sounds -> x64\Release\sounds
-	select x64 Release; unload/remove cepstral
-	build -> works fine
-	select x64 Setup
-	build -> works fine

I have NOT built a w32/x86 version in these steps -- maybe that would have pre-loaded the sound files in the "expected" place? 
I'm an entire newbie at this Windoz lark, so I'd appreciate some sanity checking.
Q:  Is this a build file "feature" or have I just not setup VS2010 properly?

all the best,
  Lawrence

From mike at jerris.com  Thu Sep  6 17:21:40 2012
From: mike at jerris.com (Michael Jerris)
Date: Thu, 6 Sep 2012 09:21:40 -0400
Subject: [Freeswitch-users] Call duration > 0 while hangup cause is not
	NORMAL_CLEARING
In-Reply-To: <CACak_yzbdE6ifgvdBPBDp78zPD7OHj5tPBzjT2XOiE1pLk5r3g@mail.gmail.com>
References: <CACak_yyyHrAR1ymkXW=Hq_s=93802Gm4ue1gh3mWvUpcMFXSDQ@mail.gmail.com>
	<CAFiqYum8tgGJHPdPgczeTb=S1zaVj9hyLXZCG1CCG0PLz-gdcw@mail.gmail.com>
	<CACc3D0UnTprOEuqhF4_=VcOzPGCiLSsMf536AN+RKY+N-Q_JbQ@mail.gmail.com>
	<CACak_ywyMXaRRjKQ-tvDjPWWK84ZVd3K+nqywypWt_saU_vijg@mail.gmail.com>
	<-6205500302962635460@unknownmsgid>
	<CACak_yzbdE6ifgvdBPBDp78zPD7OHj5tPBzjT2XOiE1pLk5r3g@mail.gmail.com>
Message-ID: <91218983-E30C-4714-B6C3-761B0594B16B@jerris.com>

if continue_on_fail is false, I don't think it will be rolling over to additional gateways.

On Sep 5, 2012, at 10:03 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:

> 
> We do bridge multiple gateways and capture failed gateways but the hangup cause shouldn't be the from the last tried gateway? That last gateway should have a good call because continue_on_fail = false. If the call fails 1st gw, the hangup cause in the cdr should come from the last tried gw rather than the 1st one.
> 
> -B
> 
> On Thu, Sep 6, 2012 at 4:51 AM, Dorn DJBinter <djbinter at gmail.com> wrote:
> Do you have call failover when you bridge?  It could be causes from the 1st gateway that failed, but your call continued to complete on the next one.
> 
> Sent from my iPad
> 
> On Sep 5, 2012, at 6:47 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
> 
>> I don't have the SIP traces right now, but I see causes like MEDIA_TIMEOUT, DESTINATION_OUT_OF_ORDER , NORMAL_UNSPECIFIED , NETWORK_OUT_OF_ORDER  and RECOVERY_ON_TIMER_EXPIRE  and the calls were still good. We are not talking about a few, they were around 50K + calls for such reasons in 24 hours. I'll try to capture SIP for any new calls. Was just curious what's going on.
>> 
>> -B
>> 
>> On Wed, Sep 5, 2012 at 6:51 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:
>> You can experience at least the following hangup causes from  FreeSWITCH  itself:
>> ALLOTTED_TIMEOUT - when you end the call by timer
>> MEDIA_TIMEOUT - when there is no RTP for some period (I would assume it can happen if server lost network connection for some time)
>> ATTENDED_TRANSFER - not sure when it will happen, but I saw this in CDR for an answered call
>> 
>> Case with media timeout is a very special one. In this case user might have talked only for 60 seconds, but timeout happened after 1800 seconds (or how is configured). You have no way to know how much time exactly the clients have talked, because in the CDR you will see 1800 seconds. In this case I think it will be fair to bill just for 1 minute or so and accept your expenses.
>> 
>> I never saw other hangup cases for answered calls from FreeSWITCH itself.
>> 
>> Regards,
>> Vitalie
>> 
>> 
>> 
>> 
>> 2012/9/5 Steven Ayre <steveayre at gmail.com>
>> Do you have SIP traces for the call? If you're bridging then it's
>> possible for the remote end to hangup the answered call with an ISDN
>> clearing cause other than NORMAL_* (just like they can in ISDN
>> signalling). For billing purposes I suggest you bill anything where
>> billsec>0 rather than only NORMAL_CLEARING calls.
>> 
>> -Steve
>> 
>> 
>> 
>> On 4 September 2012 19:23, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
>> > Hi, I am seeing a lot of calls with hangup cause/Enumeration is not
>> > NORMAL_CLEARING while the calls still have a valid duration and answered
>> > time. For a typical billing scenario, calls are billed if (answered time)
>> > billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the enumeration
>> > would be other than normal and while the calls would be still a valid
>> > answered call?
>> >
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From jeff at jefflenk.com  Thu Sep  6 17:44:03 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 6 Sep 2012 06:44:03 -0700 (PDT)
Subject: [Freeswitch-users] 1.2 stable x64 installer build issue?
In-Reply-To: <72CE317B-F7FB-4E9D-A365-B8EDC80045B1@insensate.co.uk>
References: <72CE317B-F7FB-4E9D-A365-B8EDC80045B1@insensate.co.uk>
Message-ID: <1346939043338-7582609.post@n2.nabble.com>

At this time the installer is designed to require the 32 bit version to be
run first which satisfies the sound file requirements. If you would like to
contribute a patch that address this I would be happy to except it. The
patch would need to support both 32 and 64 builds.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/1-2-stable-x64-installer-build-issue-tp7582607p7582609.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lconroy at insensate.co.uk  Thu Sep  6 18:28:54 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Thu, 6 Sep 2012 15:28:54 +0100
Subject: [Freeswitch-users] 1.2 stable x64 installer build issue?
In-Reply-To: <1346939043338-7582609.post@n2.nabble.com>
References: <72CE317B-F7FB-4E9D-A365-B8EDC80045B1@insensate.co.uk>
	<1346939043338-7582609.post@n2.nabble.com>
Message-ID: <7A5721BA-C907-4906-A7F5-1BFD879438AC@insensate.co.uk>

Hi Jeff, folks,
Many thanks.
 OK -- as I said, I'm a newbie at Windoz, so a full multi-architecture VS2010/wix patch is beyond me :(.
I will, however update the Windows installation Wiki page with your quote if I can --
I hadn't seen any doc on this & a tweaked version of your 1st sentence is all that's needed.

all the best,
  Lawrence
 
On 6 Sep 2012, at 14:44, Jeff Lenk wrote:
> At this time the installer is designed to require the 32 bit version to be
> run first which satisfies the sound file requirements. If you would like to
> contribute a patch that address this I would be happy to accept it. The
> patch would need to support both 32 and 64 builds.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/1-2-stable-x64-installer-build-issue-tp7582607p7582609.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.



From lists at kavun.ch  Thu Sep  6 18:38:44 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 10:38:44 -0400
Subject: [Freeswitch-users] att_xfer limitations
Message-ID: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>

Hi guys,

I just briefly tried the att_xfer app and it seems very limited to me. It seems to be either using a gateway or the db, but it never hits the dialplan at any point.
I love the extra add-ons you guys built, especially the option to conference before finalizing the transfer, but I think the app should be revised at some point to support a more dialplan oriented approach.
Now, when a call comes in, I can att_xfer to a SIP user, but can't call out a cellphone using my dialplan logic with my multiple outbound providers.

I encourage you to give it a look and see if it's worth an upgrade.

Best as always,
Emrah

From matt.putnam at lightspar.com  Thu Sep  6 18:42:25 2012
From: matt.putnam at lightspar.com (Matt Putnam)
Date: Thu, 6 Sep 2012 09:42:25 -0500
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
Message-ID: <9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>

Thanks for the Response Mike I am able to get the trunk registered the real issue I guess is passing calls to that trunk. As an example I have freeswitch1 for customers which has a trunk lightspar1 registered to my gateway freeswitch box. When a call is placed to a DID that is associated to freeswitch1 I can see the call in the logs but the destination number is lightspar1 not the original DID that was called. Is there a way to set the from field to use the DID called instead of the trunk name so instead of lightspar1 at blah it would send the invite with NPANXXXXXX at blah?

Thanks,
Matt Putnam
matt.putnam at lightspar.com

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, September 05, 2012 6:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch

Hi Matt,

Welcome to FreeSWITCH!

If you want something to register with FreeSWITCH then simply add an entry in the user directory. Whether it's a "user" or a "trunk" really doesn't matter - it's just a SIP registration. Look in conf/directory/default/1000.xml (if using the example "vanilla" configuration) for a taste of what you need. Let us know if you have any other questions or join us in #freeswitch on irc.freenode.net<http://irc.freenode.net>.

-MC (IRC:mercutioviz)
On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
This may have been asked before but my research hasn't turned up any results. I am currently testing freeswitch as a replacement to our current asterisk gateways. I think I have the basics of ip based trunks down but I am running into an issue with trunks registering to the freeswitch box. In asterisk it was a fairly simple process of giving the trunk a username and password and setting it to dynamic to get it to register. What would be the equivalent in freeswitch that would allow me to have customer trunks register to the system?


Thanks,
Matt Putnam


_________________________________________________________________________
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

________________________________
No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12
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From msc at freeswitch.org  Thu Sep  6 19:23:42 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Sep 2012 08:23:42 -0700
Subject: [Freeswitch-users] att_xfer limitations
In-Reply-To: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
References: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
Message-ID: <CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>

What kind of phone are you using? The reason I ask is that most desk phones
have real transfer keys that do all of this. If you're stuck using
something like x-lite then yeah, I can see the dilemma. In all honesty, the
att_xfer app wasn't designed as the be all end all of call transfers. If
you are in a scenario where your only option is to use att_xfer then I
recommend solving *that *problem. In the long run it will be much better
for you.

-MC

P.S. - It was nice having you on the conf call yesterday! Hope you can join
more calls. :)

On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:

> Hi guys,
>
> I just briefly tried the att_xfer app and it seems very limited to me. It
> seems to be either using a gateway or the db, but it never hits the
> dialplan at any point.
> I love the extra add-ons you guys built, especially the option to
> conference before finalizing the transfer, but I think the app should be
> revised at some point to support a more dialplan oriented approach.
> Now, when a call comes in, I can att_xfer to a SIP user, but can't call
> out a cellphone using my dialplan logic with my multiple outbound providers.
>
> I encourage you to give it a look and see if it's worth an upgrade.
>
> Best as always,
> 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 philq at qsystemsengineering.com  Thu Sep  6 19:29:40 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Thu, 06 Sep 2012 11:29:40 -0400
Subject: [Freeswitch-users] Far-end NAT traversal and anti-tromboning with FS
Message-ID: <00e301cd8c44$71a86ae0$54f940a0$@com>

Ah, the joys of NAT.

Is FS' NAT functionality supposed to be able to detect and handle far-end
NAT traversal?

I have the following scenario:
Endpoints on a LAN connected via the Internet to FS though NAT
FS --> Internet --> NAT firewall --> Endpoints

Proxying the media works fine but when attempting to bypass media between
endpoints all on the same LAN, I would expect FS to negotiate media between
them using their internal IP addresses.  So if IP phone A at 10.0.0.101
wants to talk to IP phone B at 10.0.0.102, FS should set up the media
streams to go directly between them using those internal addresses.
However, that is not the behavior I'm seeing.  Instead, FS is trying to set
up the media between them using their external IP addresses which results in
no audio.

Is there a way to get FS to handle this properly?  I'm pretty sure that
OpenSIPS' can detect and handle this scenario and I'm hoping that FS can as
well, without having to put OpenSIPS in front of it.  Is this just a
configuration issue?

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 msc at freeswitch.org  Thu Sep  6 19:42:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Sep 2012 08:42:58 -0700
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
Message-ID: <CAKzWOxXRD5F1_TXxr2j30StEGbsEVZjvn6WcNYaaMa-6JLv0bQ@mail.gmail.com>

Matt,

Question: do you even need a registration? The reason I ask is that you can
send calls to a server without having an actual registration. When you have
a registration then you can employ the "sofia_contact user at domain" API to
get a Sofia dialstring. However, if you know the user and domain/IP address
then you can just send a call there directly:

<action application="bridge" data="sofia/internal/5551212 at lightspar1"/>

I think your real challenge is handling the inbound call on the freeswitch1
box. That's all dialplan. Look in the example configs:
conf/dialplan/public/00_inbound_did.xml has an example of routing a call to
DID 5551212; the example extension will send the inbound call to extension
1000. You just need to copy that sample extension and set it to match your
DID and then bridge the call to your trunk.

-MC

On Thu, Sep 6, 2012 at 7:42 AM, Matt Putnam <matt.putnam at lightspar.com>wrote:

> Thanks for the Response Mike I am able to get the trunk registered the
> real issue I guess is passing calls to that trunk. As an example I have
> freeswitch1 for customers which has a trunk lightspar1 registered to my
> gateway freeswitch box. When a call is placed to a DID that is associated
> to freeswitch1 I can see the call in the logs but the destination number is
> lightspar1 not the original DID that was called. Is there a way to set the
> from field to use the DID called instead of the trunk name so instead of
> lightspar1 at blah it would send the invite with NPANXXXXXX at blah?****
>
> ** **
>
> Thanks,****
>
> Matt Putnam****
>
> matt.putnam at lightspar.com****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, September 05, 2012 6:49 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch****
>
> ** **
>
> Hi Matt,
>
> Welcome to FreeSWITCH!
>
> If you want something to register with FreeSWITCH then simply add an entry
> in the user directory. Whether it's a "user" or a "trunk" really doesn't
> matter - it's just a SIP registration. Look in
> conf/directory/default/1000.xml (if using the example "vanilla"
> configuration) for a taste of what you need. Let us know if you have any
> other questions or join us in #freeswitch on irc.freenode.net.
>
> -MC (IRC:mercutioviz)****
>
> On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com>
> wrote:****
>
> This may have been asked before but my research hasn?t turned up any
> results. I am currently testing freeswitch as a replacement to our current
> asterisk gateways. I think I have the basics of ip based trunks down but I
> am running into an issue with trunks registering to the freeswitch box. In
> asterisk it was a fairly simple process of giving the trunk a username and
> password and setting it to dynamic to get it to register. What would be the
> equivalent in freeswitch that would allow me to have customer trunks
> register to the system? ****
>
>  ****
>
>  ****
>
> Thanks,****
>
> Matt Putnam****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> ****
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12*
> ***
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lists at kavun.ch  Thu Sep  6 19:46:54 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 11:46:54 -0400
Subject: [Freeswitch-users] att_xfer limitations
In-Reply-To: <CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>
References: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
	<CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>
Message-ID: <D7E5118B-13F6-44F5-899E-B59225857F32@kavun.ch>

Hi Michael,

It was great to hear you as well, will try to be a regular. :)

I use a Polycom and it does have all the capabilities I need, but I wanted to have the PBX side working for a couple of scenarios where I thought I might need it.
E.g.: pick up a call on your cellphone, walk into the office, att_xfer the call from your cell to your desk and continue the conversation. The reason of using att_xfer is to make sure to release the call when the line is properly established and prevent calls from landing in voicemail inadvertently.

And on a final note, I never had any issue transferring calls from my SIP phones, but for some reason I find the Polycom way somewhat cumbersome and repelling. 

Best,
Emrah
On Sep 6, 2012, at 11:23 AM, Michael Collins <msc at freeswitch.org> wrote:

> What kind of phone are you using? The reason I ask is that most desk phones have real transfer keys that do all of this. If you're stuck using something like x-lite then yeah, I can see the dilemma. In all honesty, the att_xfer app wasn't designed as the be all end all of call transfers. If you are in a scenario where your only option is to use att_xfer then I recommend solving that problem. In the long run it will be much better for you.
> 
> -MC
> 
> P.S. - It was nice having you on the conf call yesterday! Hope you can join more calls. :)
> 
> On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:
> Hi guys,
> 
> I just briefly tried the att_xfer app and it seems very limited to me. It seems to be either using a gateway or the db, but it never hits the dialplan at any point.
> I love the extra add-ons you guys built, especially the option to conference before finalizing the transfer, but I think the app should be revised at some point to support a more dialplan oriented approach.
> Now, when a call comes in, I can att_xfer to a SIP user, but can't call out a cellphone using my dialplan logic with my multiple outbound providers.
> 
> I encourage you to give it a look and see if it's worth an upgrade.
> 
> Best as always,
> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Sep  6 19:48:29 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 6 Sep 2012 11:48:29 -0400
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
Message-ID: <CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>

I'm confused. I've read your message about three times, still don't
completely understand.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 6, 2012 10:44 AM, "Matt Putnam" <matt.putnam at lightspar.com> wrote:

> Thanks for the Response Mike I am able to get the trunk registered the
> real issue I guess is passing calls to that trunk. As an example I have
> freeswitch1 for customers which has a trunk lightspar1 registered to my
> gateway freeswitch box. When a call is placed to a DID that is associated
> to freeswitch1 I can see the call in the logs but the destination number is
> lightspar1 not the original DID that was called. Is there a way to set the
> from field to use the DID called instead of the trunk name so instead of
> lightspar1 at blah it would send the invite with NPANXXXXXX at blah?****
>
> ** **
>
> Thanks,****
>
> Matt Putnam****
>
> matt.putnam at lightspar.com****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, September 05, 2012 6:49 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch****
>
> ** **
>
> Hi Matt,
>
> Welcome to FreeSWITCH!
>
> If you want something to register with FreeSWITCH then simply add an entry
> in the user directory. Whether it's a "user" or a "trunk" really doesn't
> matter - it's just a SIP registration. Look in
> conf/directory/default/1000.xml (if using the example "vanilla"
> configuration) for a taste of what you need. Let us know if you have any
> other questions or join us in #freeswitch on irc.freenode.net.
>
> -MC (IRC:mercutioviz)****
>
> On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com>
> wrote:****
>
> This may have been asked before but my research hasn?t turned up any
> results. I am currently testing freeswitch as a replacement to our current
> asterisk gateways. I think I have the basics of ip based trunks down but I
> am running into an issue with trunks registering to the freeswitch box. In
> asterisk it was a fairly simple process of giving the trunk a username and
> password and setting it to dynamic to get it to register. What would be the
> equivalent in freeswitch that would allow me to have customer trunks
> register to the system? ****
>
>  ****
>
>  ****
>
> Thanks,****
>
> Matt Putnam****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> ****
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12*
> ***
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep  6 19:50:21 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 11:50:21 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <CAKzWOxVePFHNGHjWVKcPgKheHbtMu=715s5vaaoNN8B8Z63jNw@mail.gmail.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<3BFA0C81-168F-441D-9D55-D69DB58D8436@kavun.ch>
	<CAKzWOxVePFHNGHjWVKcPgKheHbtMu=715s5vaaoNN8B8Z63jNw@mail.gmail.com>
Message-ID: <DA34A0E3-424A-4F81-A28D-1C9762F3216B@kavun.ch>

I will be preaching around to have more people pull their heads out of their ASterisk. Love it!


On Sep 4, 2012, at 4:04 PM, Michael Collins <msc at freeswitch.org> wrote:

> 
> It feels as if I underwent a complete telephony mindset upgrade ever since I traded Asterisk for FS. Thank you for making it happen.
> 
> This process is known as pulling your head out of your Asterisk...
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Sep  6 19:57:22 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 11:57:22 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
Message-ID: <38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>

Hi Anthony,

The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference.
Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.

Best,
Emrah
On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Oh,
> 
> And you should try to avoid doing an attended transfer to
> one-legged-calls like conference or ivr, blind transfers work better
> for this because calls to apps are not bridged and the concept of
> transferring becomes confusing.  The other alternative is to bridge to
> the conference by looping the call over loopback or calling to the
> same box on sip so there is a true bridge.  But blind transferring is
> the best solution.
> 
> 
> On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> you are missing some of the sip and make sure its GIT HEAD (this
>> should be on jira)
>> 
>> sofia global siptrace on
>> 
>> 
>> 
>> On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
>>> Hi MC,
>>> 
>>> Thanks a bunch for your reply, sorry for the delay.
>>> Here are my logs:
>>> http://pastebin.freeswitch.org/19831
>>> 
>>> Any idea would be greatly appreciated.
>>> 
>>> Best,
>>> Emrah
>>> 
>>> 
>>> On Aug 28, 2012, at 10:56 AM, Michael Collins <msc at freeswitch.org> wrote:
>>> 
>>>> 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
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> --
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Sep  6 20:21:17 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 12:21:17 -0400
Subject: [Freeswitch-users] Best SIP phone?
Message-ID: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>

Hi all,

I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.

1. It's touch screen and I'm blind.
2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.

I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.

What can you recommend that is stable, versatile, open and good quality? 

Cheers,
Emrah

From yehavi.bourvine at gmail.com  Thu Sep  6 20:33:01 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 6 Sep 2012 19:33:01 +0300
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
Message-ID: <CAAtWKeEZzXoOb6PNKTF8md5juSUScWFR4vEBUcViJtcRcRfyCQ@mail.gmail.com>

Why won't you use the other models of Polycom which are not touch screen?
Like 450, 550, 650.

                      __yehavi:

2012/9/6 Emrah <lists at kavun.ch>

> Hi all,
>
> I have tried many IP phones and nothing comes close to the audio quality
> of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't
> suit me entirely.
>
> 1. It's touch screen and I'm blind.
> 2. It's always a pain to restart the phone anytime you make a change in
> the provisioning configs. It's great in a corporate environment, but not so
> much for experimenting.
> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial
> out from your computer's address book.
>
> I tried Snom a couple years ago and the audio quality was not good, with
> plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much
> out of the question because of their closed configs and Linksys is pretty
> low end if my recollection is correct.
>
> What can you recommend that is stable, versatile, open and good quality?
>
> Cheers,
> 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 matt.putnam at lightspar.com  Thu Sep  6 20:59:53 2012
From: matt.putnam at lightspar.com (Matt Putnam)
Date: Thu, 6 Sep 2012 11:59:53 -0500
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <CAKzWOxXRD5F1_TXxr2j30StEGbsEVZjvn6WcNYaaMa-6JLv0bQ@mail.gmail.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKzWOxXRD5F1_TXxr2j30StEGbsEVZjvn6WcNYaaMa-6JLv0bQ@mail.gmail.com>
Message-ID: <9757304EEA8CE74494B6D21E63DA005212AE44D13B@MBX23.exg5.exghost.com>

Registration is not necessary in most situations but there are a few situation where a customer is not on a static ip so registration would be necessary.

Thanks,
Matt Putnam
matt.putnam at lightspar.com

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, September 06, 2012 10:43 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch

Matt,

Question: do you even need a registration? The reason I ask is that you can send calls to a server without having an actual registration. When you have a registration then you can employ the "sofia_contact user at domain" API to get a Sofia dialstring. However, if you know the user and domain/IP address then you can just send a call there directly:

<action application="bridge" data="sofia/internal/5551212 at lightspar1"/>

I think your real challenge is handling the inbound call on the freeswitch1 box. That's all dialplan. Look in the example configs: conf/dialplan/public/00_inbound_did.xml has an example of routing a call to DID 5551212; the example extension will send the inbound call to extension 1000. You just need to copy that sample extension and set it to match your DID and then bridge the call to your trunk.

-MC
On Thu, Sep 6, 2012 at 7:42 AM, Matt Putnam <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
Thanks for the Response Mike I am able to get the trunk registered the real issue I guess is passing calls to that trunk. As an example I have freeswitch1 for customers which has a trunk lightspar1 registered to my gateway freeswitch box. When a call is placed to a DID that is associated to freeswitch1 I can see the call in the logs but the destination number is lightspar1 not the original DID that was called. Is there a way to set the from field to use the DID called instead of the trunk name so instead of lightspar1 at blah it would send the invite with NPANXXXXXX at blah?

Thanks,
Matt Putnam
matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Wednesday, September 05, 2012 6:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch

Hi Matt,

Welcome to FreeSWITCH!

If you want something to register with FreeSWITCH then simply add an entry in the user directory. Whether it's a "user" or a "trunk" really doesn't matter - it's just a SIP registration. Look in conf/directory/default/1000.xml (if using the example "vanilla" configuration) for a taste of what you need. Let us know if you have any other questions or join us in #freeswitch on irc.freenode.net<http://irc.freenode.net>.

-MC (IRC:mercutioviz)
On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
This may have been asked before but my research hasn't turned up any results. I am currently testing freeswitch as a replacement to our current asterisk gateways. I think I have the basics of ip based trunks down but I am running into an issue with trunks registering to the freeswitch box. In asterisk it was a fairly simple process of giving the trunk a username and password and setting it to dynamic to get it to register. What would be the equivalent in freeswitch that would allow me to have customer trunks register to the system?


Thanks,
Matt Putnam


_________________________________________________________________________
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
________________________________
No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12

_________________________________________________________________________
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

________________________________
No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 2012.0.2197 / Virus Database: 2437/5252 - Release Date: 09/06/12
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From lists at kavun.ch  Thu Sep  6 22:09:23 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 14:09:23 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <CAAtWKeEZzXoOb6PNKTF8md5juSUScWFR4vEBUcViJtcRcRfyCQ@mail.gmail.com>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<CAAtWKeEZzXoOb6PNKTF8md5juSUScWFR4vEBUcViJtcRcRfyCQ@mail.gmail.com>
Message-ID: <0E46F8E7-FD3C-450D-9632-8482C1D9D5FD@kavun.ch>

Thanks for your suggestion.
I've used all three of them in the past and even so, they are not as practical as other phones I've owned. Plus, the VVX 1500 supports Siren14 which we use across the company on a daily basis. Finally, video isn't a real requirement, but we tend to use it when it's available.

Cheers,
E
On Sep 6, 2012, at 12:33 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:

> Why won't you use the other models of Polycom which are not touch screen? Like 450, 550, 650.
>  
>                       __yehavi:
> 
> 2012/9/6 Emrah <lists at kavun.ch>
> Hi all,
> 
> I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.
> 
> 1. It's touch screen and I'm blind.
> 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.
> 
> I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.
> 
> What can you recommend that is stable, versatile, open and good quality?
> 
> Cheers,
> 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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jmesquita at freeswitch.org  Thu Sep  6 23:03:09 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 6 Sep 2012 16:03:09 -0300
Subject: [Freeswitch-users] att_xfer limitations
In-Reply-To: <D7E5118B-13F6-44F5-899E-B59225857F32@kavun.ch>
References: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
	<CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>
	<D7E5118B-13F6-44F5-899E-B59225857F32@kavun.ch>
Message-ID: <CABH8QX1B-4VSR7C-AoZCK3o3yyv27777JOYfEqgOud9BQVstxA@mail.gmail.com>

I do believe that there are some legit cases where att_xfer is an
application that needs to be fully functional. The most obvious case is
where analog phone are used with boards such as Sangoma, Khomp or Digium
boards. It is in this instance arguable if that needs to be implemented as
an application on the "core" so to speak on or the endpoint module.

It is not a trivial task (at least beyond my capabilities) to make some
extra things work on att_xfer such as call return.

Nonetheless, Emrah, what you are asking is doable with the current
implementation. Just use the loopback channel and make
loopback_bowout=false. The downside in this case is that the loopback
channel will be up during the entire duration of the call, but everything
else will work.

Also, this adds some extra complexity to CDR processing, but I haven't
worked on that to really know how much complexity.

I hope this works for you.

Regards,
Jo?o Mesquita



On Thu, Sep 6, 2012 at 12:46 PM, Emrah <lists at kavun.ch> wrote:

> Hi Michael,
>
> It was great to hear you as well, will try to be a regular. :)
>
> I use a Polycom and it does have all the capabilities I need, but I wanted
> to have the PBX side working for a couple of scenarios where I thought I
> might need it.
> E.g.: pick up a call on your cellphone, walk into the office, att_xfer the
> call from your cell to your desk and continue the conversation. The reason
> of using att_xfer is to make sure to release the call when the line is
> properly established and prevent calls from landing in voicemail
> inadvertently.
>
> And on a final note, I never had any issue transferring calls from my SIP
> phones, but for some reason I find the Polycom way somewhat cumbersome and
> repelling.
>
> Best,
> Emrah
> On Sep 6, 2012, at 11:23 AM, Michael Collins <msc at freeswitch.org> wrote:
>
> > What kind of phone are you using? The reason I ask is that most desk
> phones have real transfer keys that do all of this. If you're stuck using
> something like x-lite then yeah, I can see the dilemma. In all honesty, the
> att_xfer app wasn't designed as the be all end all of call transfers. If
> you are in a scenario where your only option is to use att_xfer then I
> recommend solving that problem. In the long run it will be much better for
> you.
> >
> > -MC
> >
> > P.S. - It was nice having you on the conf call yesterday! Hope you can
> join more calls. :)
> >
> > On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:
> > Hi guys,
> >
> > I just briefly tried the att_xfer app and it seems very limited to me.
> It seems to be either using a gateway or the db, but it never hits the
> dialplan at any point.
> > I love the extra add-ons you guys built, especially the option to
> conference before finalizing the transfer, but I think the app should be
> revised at some point to support a more dialplan oriented approach.
> > Now, when a call comes in, I can att_xfer to a SIP user, but can't call
> out a cellphone using my dialplan logic with my multiple outbound providers.
> >
> > I encourage you to give it a look and see if it's worth an upgrade.
> >
> > Best as always,
> > 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
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From matt.putnam at lightspar.com  Thu Sep  6 23:09:22 2012
From: matt.putnam at lightspar.com (Matt Putnam)
Date: Thu, 6 Sep 2012 14:09:22 -0500
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
Message-ID: <9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>

Sorry for the confusion I can really only equate it to asterisk as that's what our current platform is. Essentially the problem is that if I have a PBX that has a trunk that is registered to freeswitch and a call is received for a DID  that is on that trunk I am currently unable to send that DID to the PBX. When the call is received on the PBX side it looks as if the call is for the trunk instead of the DID is should be for. So in the SIP invite instead of using lightspar1 at domain it would use 5551212 at domain. In asterisk this was simply accomplished by a dial statement of (SIP/5551212 at lightspar1) what I am looking for is that equivalent in freeswitch.

Thanks,
Matt Putnam
matt.putnam at lightspar.com

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Thursday, September 06, 2012 10:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch


I'm confused. I've read your message about three times, still don't completely understand.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 6, 2012 10:44 AM, "Matt Putnam" <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
Thanks for the Response Mike I am able to get the trunk registered the real issue I guess is passing calls to that trunk. As an example I have freeswitch1 for customers which has a trunk lightspar1 registered to my gateway freeswitch box. When a call is placed to a DID that is associated to freeswitch1 I can see the call in the logs but the destination number is lightspar1 not the original DID that was called. Is there a way to set the from field to use the DID called instead of the trunk name so instead of lightspar1 at blah it would send the invite with NPANXXXXXX at blah?

Thanks,
Matt Putnam
matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Wednesday, September 05, 2012 6:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch

Hi Matt,

Welcome to FreeSWITCH!

If you want something to register with FreeSWITCH then simply add an entry in the user directory. Whether it's a "user" or a "trunk" really doesn't matter - it's just a SIP registration. Look in conf/directory/default/1000.xml (if using the example "vanilla" configuration) for a taste of what you need. Let us know if you have any other questions or join us in #freeswitch on irc.freenode.net<http://irc.freenode.net>.

-MC (IRC:mercutioviz)
On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
This may have been asked before but my research hasn't turned up any results. I am currently testing freeswitch as a replacement to our current asterisk gateways. I think I have the basics of ip based trunks down but I am running into an issue with trunks registering to the freeswitch box. In asterisk it was a fairly simple process of giving the trunk a username and password and setting it to dynamic to get it to register. What would be the equivalent in freeswitch that would allow me to have customer trunks register to the system?


Thanks,
Matt Putnam


_________________________________________________________________________
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
________________________________
No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12

_________________________________________________________________________
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
________________________________
No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 2012.0.2197 / Virus Database: 2437/5252 - Release Date: 09/06/12
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From abaci64 at gmail.com  Thu Sep  6 23:20:40 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 06 Sep 2012 15:20:40 -0400
Subject: [Freeswitch-users] effective_callee_id_name behaviour
In-Reply-To: <CAKbxfG_=OZHu7hp9K9MJDobRFCce4Dse4n1BmVubb47tYpFncw@mail.gmail.com>
References: <CAAtWKeEXaWLfXL89QcwxHCHXWE07up13mgOwUUCytRNpv9EogQ@mail.gmail.com>
	<CAKbxfG_=OZHu7hp9K9MJDobRFCce4Dse4n1BmVubb47tYpFncw@mail.gmail.com>
Message-ID: <5048F788.8040005@gmail.com>

Yehavi,
I don't see these variables documented on wiki, can you please document 
them once you get them working.
see http://wiki.freeswitch.org/wiki/Channel_Variables#Callee_ID_Related

On 9/5/2012 12:04 PM, Anthony Minessale wrote:
> {origination_callee_id_name='test user',origination_callee_id_number=5551212}
>
>
> On Wed, Sep 5, 2012 at 5:56 AM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
>> Hello,
>>
>>    Our FreeSwitch is connected to a Nortel PBX via E1-SIP gateway
>> (AudioCodes). This gateway supports the P-Asserted-ID field.
>>
>> With a vanilla configuration the following happens when a FS user calls a
>> user on the Nortel:
>> - During the ringing phase the name of the callee is "outbound call"
>> - When the other side answers the name of the callee is set to the one sent
>> from the Nortel.
>>
>> In order to have the name also during the ringing phase, I set calle_id_name
>> and effective_callee_id_name.
>> After doing so, the name is not changed after the remote user answers. I
>> need it to change to the name sent
>> from the Nortel, as it shows the name of the one who actually answered the
>> phone.
>>
>> I also tried adding ignore_display_updates=false, but it didn't change the
>> behaviour.
>>
>> Any idea how to do it?
>>
>>                          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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



From anthony.minessale at gmail.com  Thu Sep  6 23:23:00 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Sep 2012 14:23:00 -0500
Subject: [Freeswitch-users] att_xfer limitations
In-Reply-To: <CABH8QX1B-4VSR7C-AoZCK3o3yyv27777JOYfEqgOud9BQVstxA@mail.gmail.com>
References: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
	<CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>
	<D7E5118B-13F6-44F5-899E-B59225857F32@kavun.ch>
	<CABH8QX1B-4VSR7C-AoZCK3o3yyv27777JOYfEqgOud9BQVstxA@mail.gmail.com>
Message-ID: <CAKbxfG8dexu-3yJEYG8qjcunAWi8wzsqR+mGjTRRiJgvHfzS1w@mail.gmail.com>

Emrah,

I suggest you further study the application as a whole before making
determinations on where improvements need to be made.
You may need to think a bit more "out of the box" if you want to
be successful.



On Thu, Sep 6, 2012 at 2:03 PM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:

> I do believe that there are some legit cases where att_xfer is an
> application that needs to be fully functional. The most obvious case is
> where analog phone are used with boards such as Sangoma, Khomp or Digium
> boards. It is in this instance arguable if that needs to be implemented as
> an application on the "core" so to speak on or the endpoint module.
>
> It is not a trivial task (at least beyond my capabilities) to make some
> extra things work on att_xfer such as call return.
>
> Nonetheless, Emrah, what you are asking is doable with the current
> implementation. Just use the loopback channel and make
> loopback_bowout=false. The downside in this case is that the loopback
> channel will be up during the entire duration of the call, but everything
> else will work.
>
> Also, this adds some extra complexity to CDR processing, but I haven't
> worked on that to really know how much complexity.
>
> I hope this works for you.
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Thu, Sep 6, 2012 at 12:46 PM, Emrah <lists at kavun.ch> wrote:
>
>> Hi Michael,
>>
>> It was great to hear you as well, will try to be a regular. :)
>>
>> I use a Polycom and it does have all the capabilities I need, but I
>> wanted to have the PBX side working for a couple of scenarios where I
>> thought I might need it.
>> E.g.: pick up a call on your cellphone, walk into the office, att_xfer
>> the call from your cell to your desk and continue the conversation. The
>> reason of using att_xfer is to make sure to release the call when the line
>> is properly established and prevent calls from landing in voicemail
>> inadvertently.
>>
>> And on a final note, I never had any issue transferring calls from my SIP
>> phones, but for some reason I find the Polycom way somewhat cumbersome and
>> repelling.
>>
>> Best,
>> Emrah
>> On Sep 6, 2012, at 11:23 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> > What kind of phone are you using? The reason I ask is that most desk
>> phones have real transfer keys that do all of this. If you're stuck using
>> something like x-lite then yeah, I can see the dilemma. In all honesty, the
>> att_xfer app wasn't designed as the be all end all of call transfers. If
>> you are in a scenario where your only option is to use att_xfer then I
>> recommend solving that problem. In the long run it will be much better for
>> you.
>> >
>> > -MC
>> >
>> > P.S. - It was nice having you on the conf call yesterday! Hope you can
>> join more calls. :)
>> >
>> > On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:
>> > Hi guys,
>> >
>> > I just briefly tried the att_xfer app and it seems very limited to me.
>> It seems to be either using a gateway or the db, but it never hits the
>> dialplan at any point.
>> > I love the extra add-ons you guys built, especially the option to
>> conference before finalizing the transfer, but I think the app should be
>> revised at some point to support a more dialplan oriented approach.
>> > Now, when a call comes in, I can att_xfer to a SIP user, but can't call
>> out a cellphone using my dialplan logic with my multiple outbound providers.
>> >
>> > I encourage you to give it a look and see if it's worth an upgrade.
>> >
>> > Best as always,
>> > 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
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike.burlingame at me.com  Thu Sep  6 23:50:34 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Thu, 06 Sep 2012 12:50:34 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <927793A3-06D4-4AFD-8AEC-69745E2BC775@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
Message-ID: <FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>

After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.

Thanks

On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame at me.com> wrote:

> Looks much much better Thank you -- Now to conduct more testing 
> 
> 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-Leg:5060, with session description
>  0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying
>  0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>  0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>  0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>  0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
>  0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>  5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>  5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description
>  5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
>  5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
> 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK
> 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
> 
> 
> 
> On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> ok,
>> 
>> update one more time, if it still does not work just go right to jira
>> with the latest (not before today's changes)
>> 
>> 
>> On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
>>> 
>>> commit d45db898339e1b2212f5befff1af714abcec034f
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> Date:   Wed Sep 5 13:11:32 2012 -0500
>>> 
>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>>> 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>> 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>> 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>> 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>> 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
>>> 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>>> 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>>> 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>>> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>> 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
>>> 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
>>> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
>>> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>>> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>>> 
>>> 
>>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> 
>>>> update and try again, if it still doesn't work open a jira
>>>> 
>>>> 
>>>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>> as promised here is the update testing and enabling
>>>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
>>>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
>>>>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>>>>> 
>>>>> I can open a JIRA case on this and provide the console log file / PCAP's ect
>>>>> if that would help
>>>>> 
>>>>> 
>>>>> Call Flow with out {sip_wait_for_aleg_ack=true}
>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>>>> sip:+13605551212 at A-LEG:5060, with session description
>>>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>>>> sip:13605551212 at B-Leg, with session description
>>>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>>>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>>> 
>>>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>>>> sip:+13605551212 at A-LEG:5060, with session description
>>>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>>>> sip:13605551212 at B-Leg, with session description
>>>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>>>> description
>>>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>>>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>>> 
>>>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>> 
>>>>> No worries I will be out this weekend for the long weekend I will work on
>>>>> getting the test box upgraded and a test case setup on Tuesday I will report
>>>>> back the results mid to late next week and provided everything works as I
>>>>> hope it will I will be happy to pay the Wiki tax :)
>>>>> 
>>>>> 
>>>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>>> 
>>>>> 
>>>>> 
>>>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
>>>>> wrote:
>>>>>> 
>>>>>> Cool I will nail that up on my test box and see if that works
>>>>>> 
>>>>> Please report back on whether it works or not and then be prepared to pay
>>>>> the wiki tax. :) I'll be glad to assist with getting this documented
>>>>> although I think you're in the best position to give that documentation some
>>>>> real-world context.
>>>>> 
>>>>> -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
>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Sep  7 00:03:40 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 06 Sep 2012 16:03:40 -0400
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
Message-ID: <17431.1346961820@ccs.covici.com>

You might try to use the info app to see if there is a variable with the
did number -- I have seen trunks do that myself.

Matt Putnam <matt.putnam at lightspar.com> wrote:

> Sorry for the confusion I can really only equate it to asterisk as that's what our current platform is. Essentially the problem is that if I have a PBX that has a trunk that is registered to freeswitch and a call is received for a DID  that is on that trunk I am currently unable to send that DID to the PBX. When the call is received on the PBX side it looks as if the call is for the trunk instead of the DID is should be for. So in the SIP invite instead of using lightspar1 at domain it would use 5551212 at domain. In asterisk this was simply accomplished by a dial statement of (SIP/5551212 at lightspar1) what I am looking for is that equivalent in freeswitch.
> 
> Thanks,
> Matt Putnam
> matt.putnam at lightspar.com
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
> Sent: Thursday, September 06, 2012 10:48 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch
> 
> 
> I'm confused. I've read your message about three times, still don't completely understand.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> On Sep 6, 2012 10:44 AM, "Matt Putnam" <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
> Thanks for the Response Mike I am able to get the trunk registered the real issue I guess is passing calls to that trunk. As an example I have freeswitch1 for customers which has a trunk lightspar1 registered to my gateway freeswitch box. When a call is placed to a DID that is associated to freeswitch1 I can see the call in the logs but the destination number is lightspar1 not the original DID that was called. Is there a way to set the from field to use the DID called instead of the trunk name so instead of lightspar1 at blah it would send the invite with NPANXXXXXX at blah?
> 
> Thanks,
> Matt Putnam
> matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>
> 
> From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
> Sent: Wednesday, September 05, 2012 6:49 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch
> 
> Hi Matt,
> 
> Welcome to FreeSWITCH!
> 
> If you want something to register with FreeSWITCH then simply add an entry in the user directory. Whether it's a "user" or a "trunk" really doesn't matter - it's just a SIP registration. Look in conf/directory/default/1000.xml (if using the example "vanilla" configuration) for a taste of what you need. Let us know if you have any other questions or join us in #freeswitch on irc.freenode.net<http://irc.freenode.net>.
> 
> -MC (IRC:mercutioviz)
> On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
> This may have been asked before but my research hasn't turned up any results. I am currently testing freeswitch as a replacement to our current asterisk gateways. I think I have the basics of ip based trunks down but I am running into an issue with trunks registering to the freeswitch box. In asterisk it was a fairly simple process of giving the trunk a username and password and setting it to dynamic to get it to register. What would be the equivalent in freeswitch that would allow me to have customer trunks register to the system?
> 
> 
> Thanks,
> Matt Putnam
> 
> 
> _________________________________________________________________________
> 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
> ________________________________
> No virus found in this message.
> Checked by AVG - www.avg.com<http://www.avg.com>
> Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12
> 
> _________________________________________________________________________
> 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
> ________________________________
> No virus found in this message.
> Checked by AVG - www.avg.com<http://www.avg.com>
> Version: 2012.0.2197 / Virus Database: 2437/5252 - Release Date: 09/06/12
> 
> ----------------------------------------------------
> 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 lists at kavun.ch  Fri Sep  7 00:09:46 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 16:09:46 -0400
Subject: [Freeswitch-users] att_xfer limitations
In-Reply-To: <CAKbxfG8dexu-3yJEYG8qjcunAWi8wzsqR+mGjTRRiJgvHfzS1w@mail.gmail.com>
References: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
	<CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>
	<D7E5118B-13F6-44F5-899E-B59225857F32@kavun.ch>
	<CABH8QX1B-4VSR7C-AoZCK3o3yyv27777JOYfEqgOud9BQVstxA@mail.gmail.com>
	<CAKbxfG8dexu-3yJEYG8qjcunAWi8wzsqR+mGjTRRiJgvHfzS1w@mail.gmail.com>
Message-ID: <015D0E1F-03A8-4452-B0F0-8657C273B5E4@kavun.ch>

Dear Anthony,

You're message did not come across very pleasantly, but as it is often the case in writing, sometimes the receiver does not perceive things as they were meant originally. I'll disregard the judgmental portion of your email.

When you refer to the application as a whole, should I assume FreeSWITCH or ATT_XFER? 
When you mean thinking outside the box, am I to assume creating work arounds (e.g.: by using loopback calls?) I don't find that quite sexy to be honest.

My previous message never demanded or even asked for an upgrade of a functionality. I stated my thoughts and encouraged the developers to consider my approach and see whether it was worth a revision. I obviously did not mean to attack your work in any way.

So here are my 2 cents:
The att_xfer feature has been implemented for a certain reason. I've seen it working differently on Asterisk where the dialplan is used all the time. Based on that, and as a personal thought worth mentioning, I suggested a similar approach. Leaving the feature as it is, and based on the limited knowledge I currently have on FreeSWITCH, it makes it unusable in any of my use cases.

I divorced my Asterisk boxes and made a commitment to FS. You've built something amazing and I will always try to be constructive where I can.

Best to you Anthony, and again congrats for the super initiative you took by creating FS.

Emrah


On Sep 6, 2012, at 3:23 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Emrah,
> 
> I suggest you further study the application as a whole before making determinations on where improvements need to be made.
> You may need to think a bit more "out of the box" if you want to be successful.
> 
> 
> 
> On Thu, Sep 6, 2012 at 2:03 PM, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
> I do believe that there are some legit cases where att_xfer is an application that needs to be fully functional. The most obvious case is where analog phone are used with boards such as Sangoma, Khomp or Digium boards. It is in this instance arguable if that needs to be implemented as an application on the "core" so to speak on or the endpoint module.
> 
> It is not a trivial task (at least beyond my capabilities) to make some extra things work on att_xfer such as call return.
> 
> Nonetheless, Emrah, what you are asking is doable with the current implementation. Just use the loopback channel and make loopback_bowout=false. The downside in this case is that the loopback channel will be up during the entire duration of the call, but everything else will work.
> 
> Also, this adds some extra complexity to CDR processing, but I haven't worked on that to really know how much complexity.
> 
> I hope this works for you.
> 
> Regards,
> Jo?o Mesquita
> 
> 
> 
> 
> On Thu, Sep 6, 2012 at 12:46 PM, Emrah <lists at kavun.ch> wrote:
> Hi Michael,
> 
> It was great to hear you as well, will try to be a regular. :)
> 
> I use a Polycom and it does have all the capabilities I need, but I wanted to have the PBX side working for a couple of scenarios where I thought I might need it.
> E.g.: pick up a call on your cellphone, walk into the office, att_xfer the call from your cell to your desk and continue the conversation. The reason of using att_xfer is to make sure to release the call when the line is properly established and prevent calls from landing in voicemail inadvertently.
> 
> And on a final note, I never had any issue transferring calls from my SIP phones, but for some reason I find the Polycom way somewhat cumbersome and repelling.
> 
> Best,
> Emrah
> On Sep 6, 2012, at 11:23 AM, Michael Collins <msc at freeswitch.org> wrote:
> 
> > What kind of phone are you using? The reason I ask is that most desk phones have real transfer keys that do all of this. If you're stuck using something like x-lite then yeah, I can see the dilemma. In all honesty, the att_xfer app wasn't designed as the be all end all of call transfers. If you are in a scenario where your only option is to use att_xfer then I recommend solving that problem. In the long run it will be much better for you.
> >
> > -MC
> >
> > P.S. - It was nice having you on the conf call yesterday! Hope you can join more calls. :)
> >
> > On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:
> > Hi guys,
> >
> > I just briefly tried the att_xfer app and it seems very limited to me. It seems to be either using a gateway or the db, but it never hits the dialplan at any point.
> > I love the extra add-ons you guys built, especially the option to conference before finalizing the transfer, but I think the app should be revised at some point to support a more dialplan oriented approach.
> > Now, when a call comes in, I can att_xfer to a SIP user, but can't call out a cellphone using my dialplan logic with my multiple outbound providers.
> >
> > I encourage you to give it a look and see if it's worth an upgrade.
> >
> > Best as always,
> > 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
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 itispip-qq at hotmail.com  Thu Sep  6 21:19:39 2012
From: itispip-qq at hotmail.com (Pip Live)
Date: Fri, 7 Sep 2012 01:19:39 +0800
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
Message-ID: <BLU0-SMTP4444D917A72D348AABEB9DE8CA80@phx.gbl>

under \freeswitch\conf\sip\default or public, modify any template suit you.
? 2012-9-6 ??6:21?"Matt Putnam" <matt.putnam at lightspar.com>???

> This may have been asked before but my research hasn?t turned up any
> results. I am currently testing freeswitch as a replacement to our current
> asterisk gateways. I think I have the basics of ip based trunks down but I
> am running into an issue with trunks registering to the freeswitch box. In
> asterisk it was a fairly simple process of giving the trunk a username and
> password and setting it to dynamic to get it to register. What would be the
> equivalent in freeswitch that would allow me to have customer trunks
> register to the system? ****
>
> ** **
>
> ** **
>
> Thanks,****
>
> Matt Putnam****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Sep  7 02:19:10 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 6 Sep 2012 15:19:10 -0700 (PDT)
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
	<38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
Message-ID: <1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>

Emrah, I think this particular use case should be quite easy to program in a Lua script.
The incoming call is handed to the script, the script originates a call to the operator, 
bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.


on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.




----- Original Message -----
> From: Emrah <lists at kavun.ch>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Thursday, September 6, 2012 5:57 PM
> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
> 
> Hi Anthony,
> 
> The reason for the attended transfer to the conference call is to announce 
> "manually" a new participant before releasing the transfer. E.g.: in 
> the scenario where you have an operator transferring calling parties in a 
> conference.
> Which of the two alternatives would be the most CPU efficient? A loopback or a 
> SIP call to the same domain? I wish I could give a hand in the code, but my 2 
> cents would be to optimize this at some point in your roadmap.
> 
> Best,
> Emrah
> On Sep 4, 2012, at 1:17 PM, Anthony Minessale 
> <anthony.minessale at gmail.com> wrote:
> 
>>  Oh,
>> 
>>  And you should try to avoid doing an attended transfer to
>>  one-legged-calls like conference or ivr, blind transfers work better
>>  for this because calls to apps are not bridged and the concept of
>>  transferring becomes confusing.? The other alternative is to bridge to
>>  the conference by looping the call over loopback or calling to the
>>  same box on sip so there is a true bridge.? But blind transferring is
>>  the best solution.
>> 
>> 
>>  On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
>>  <anthony.minessale at gmail.com> wrote:
>>>  you are missing some of the sip and make sure its GIT HEAD (this
>>>  should be on jira)
>>> 
>>>  sofia global siptrace on
>>> 
>>> 
>>> 
>>>  On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
>>>>  Hi MC,
>>>> 
>>>>  Thanks a bunch for your reply, sorry for the delay.
>>>>  Here are my logs:
>>>>  http://pastebin.freeswitch.org/19831
>>>> 
>>>>  Any idea would be greatly appreciated.
>>>> 
>>>>  Best,
>>>>  Emrah
>>>> 
>>>> 
>>>>  On Aug 28, 2012, at 10:56 AM, Michael Collins 
> <msc at freeswitch.org> wrote:
>>>> 
>>>>>  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
>>>>> 
>>>>> 
>>>>> 
> _________________________________________________________________________
>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>  consulting at freeswitch.org
>>>>>  http://www.freeswitchsolutions.com
>>>>> 
>>>>>  
>>>>>  
>>>>> 
>>>>>  Official FreeSWITCH Sites
>>>>>  http://www.freeswitch.org
>>>>>  http://wiki.freeswitch.org
>>>>>  http://www.cluecon.com
>>>>> 
>>>>>  FreeSWITCH-users mailing list
>>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
> _________________________________________________________________________
>>>>  Professional FreeSWITCH Consulting Services:
>>>>  consulting at freeswitch.org
>>>>  http://www.freeswitchsolutions.com
>>>> 
>>>>  
>>>>  
>>>> 
>>>>  Official FreeSWITCH Sites
>>>>  http://www.freeswitch.org
>>>>  http://wiki.freeswitch.org
>>>>  http://www.cluecon.com
>>>> 
>>>>  FreeSWITCH-users mailing list
>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  http://www.freeswitch.org
>>> 
>>> 
>>> 
>>>  --
>>>  Anthony Minessale II
>>> 
>>>  FreeSWITCH http://www.freeswitch.org/
>>>  ClueCon http://www.cluecon.com/
>>>  Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>>  AIM: anthm
>>>  MSN:anthony_minessale at hotmail.com
>>>  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>  IRC: irc.freenode.net #freeswitch
>>> 
>>>  FreeSWITCH Developer Conference
>>>  sip:888 at conference.freeswitch.org
>>>  googletalk:conf+888 at conference.freeswitch.org
>>>  pstn:+19193869900
>> 
>> 
>> 
>>  -- 
>>  Anthony Minessale II
>> 
>>  FreeSWITCH http://www.freeswitch.org/
>>  ClueCon http://www.cluecon.com/
>>  Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>>  AIM: anthm
>>  MSN:anthony_minessale at hotmail.com
>>  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>  IRC: irc.freenode.net #freeswitch
>> 
>>  FreeSWITCH Developer Conference
>>  sip:888 at conference.freeswitch.org
>>  googletalk:conf+888 at conference.freeswitch.org
>>  pstn:+19193869900
>> 
>>  _________________________________________________________________________
>>  Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org
>>  http://www.freeswitchsolutions.com
>> 
>>  
>>  
>> 
>>  Official FreeSWITCH Sites
>>  http://www.freeswitch.org
>>  http://wiki.freeswitch.org
>>  http://www.cluecon.com
>> 
>>  FreeSWITCH-users mailing list
>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From ssinyagin at yahoo.com  Fri Sep  7 02:32:09 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 6 Sep 2012 15:32:09 -0700 (PDT)
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
	<38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
	<1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <1346970729.17214.YahooMailNeo@web39301.mail.mud.yahoo.com>

after a second thought, I realized that you probably want to minimize your own DTMF dialing effort after the call is set up. This can be done easier:? you can build additional extensions which only work for your SIP account and connect into the conference rooms without PIN. This can be done (again) with a Lua script, or probably by manipulating the contexts. 





----- Original Message -----
> From: Stanislav Sinyagin <ssinyagin at yahoo.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Friday, September 7, 2012 12:19 AM
> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
> 
> Emrah, I think this particular use case should be quite easy to program in a Lua 
> script.
> The incoming call is handed to the script, the script originates a call to the 
> operator, 
> bridges them, then waits for operator's input, then hangs up the operator 
> and bridges the A leg into the conference.
> 
> 
> on the other side, why would anyone need a human operator in front of a 
> conference bridge? 4 or 5 digits PIN usually works fine.
> 
> 
> 
> 
> ----- Original Message -----
>>  From: Emrah <lists at kavun.ch>
>>  To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>  Cc: 
>>  Sent: Thursday, September 6, 2012 5:57 PM
>>  Subject: Re: [Freeswitch-users] Attended transfer to a conference room
>> 
>>  Hi Anthony,
>> 
>>  The reason for the attended transfer to the conference call is to announce 
>>  "manually" a new participant before releasing the transfer. E.g.: 
> in 
>>  the scenario where you have an operator transferring calling parties in a 
>>  conference.
>>  Which of the two alternatives would be the most CPU efficient? A loopback 
> or a 
>>  SIP call to the same domain? I wish I could give a hand in the code, but my 
> 2 
>>  cents would be to optimize this at some point in your roadmap.
>> 
>>  Best,
>>  Emrah
>>  On Sep 4, 2012, at 1:17 PM, Anthony Minessale 
>>  <anthony.minessale at gmail.com> wrote:
>> 
>>> ? Oh,
>>> 
>>> ? And you should try to avoid doing an attended transfer to
>>> ? one-legged-calls like conference or ivr, blind transfers work better
>>> ? for this because calls to apps are not bridged and the concept of
>>> ? transferring becomes confusing.? The other alternative is to bridge to
>>> ? the conference by looping the call over loopback or calling to the
>>> ? same box on sip so there is a true bridge.? But blind transferring is
>>> ? the best solution.
>>> 
>>> 
>>> ? On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
>>> ? <anthony.minessale at gmail.com> wrote:
>>>> ? you are missing some of the sip and make sure its GIT HEAD (this
>>>> ? should be on jira)
>>>> 
>>>> ? sofia global siptrace on
>>>> 
>>>> 
>>>> 
>>>> ? On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> 
> wrote:
>>>>> ? Hi MC,
>>>>> 
>>>>> ? Thanks a bunch for your reply, sorry for the delay.
>>>>> ? Here are my logs:
>>>>> ? http://pastebin.freeswitch.org/19831
>>>>> 
>>>>> ? Any idea would be greatly appreciated.
>>>>> 
>>>>> ? Best,
>>>>> ? Emrah
>>>>> 
>>>>> 
>>>>> ? On Aug 28, 2012, at 10:56 AM, Michael Collins 
>>  <msc at freeswitch.org> wrote:
>>>>> 
>>>>>> ? 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
>>>>>> 
>>>>>> ? 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
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> ? --
>>>>>> ? 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
>>>>>> 
>>>>>> ? 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
>>>>> 
>>>>> 
>>>>> 
>>  _________________________________________________________________________
>>>>> ? Professional FreeSWITCH Consulting Services:
>>>>> ? consulting at freeswitch.org
>>>>> ? http://www.freeswitchsolutions.com
>>>>> 
>>>>> ? 
>>>>> ? 
>>>>> 
>>>>> ? Official FreeSWITCH Sites
>>>>> ? http://www.freeswitch.org
>>>>> ? http://wiki.freeswitch.org
>>>>> ? http://www.cluecon.com
>>>>> 
>>>>> ? FreeSWITCH-users mailing list
>>>>> ? FreeSWITCH-users at lists.freeswitch.org
>>>>> ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> ? http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> ? --
>>>> ? Anthony Minessale II
>>>> 
>>>> ? FreeSWITCH http://www.freeswitch.org/
>>>> ? ClueCon http://www.cluecon.com/
>>>> ? Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>> ? AIM: anthm
>>>> ? MSN:anthony_minessale at hotmail.com
>>>> ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> ? IRC: irc.freenode.net #freeswitch
>>>> 
>>>> ? FreeSWITCH Developer Conference
>>>> ? sip:888 at conference.freeswitch.org
>>>> ? googletalk:conf+888 at conference.freeswitch.org
>>>> ? pstn:+19193869900
>>> 
>>> 
>>> 
>>> ? -- 
>>> ? Anthony Minessale II
>>> 
>>> ? FreeSWITCH http://www.freeswitch.org/
>>> ? ClueCon http://www.cluecon.com/
>>> ? Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> ? AIM: anthm
>>> ? MSN:anthony_minessale at hotmail.com
>>> ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> ? IRC: irc.freenode.net #freeswitch
>>> 
>>> ? FreeSWITCH Developer Conference
>>> ? sip:888 at conference.freeswitch.org
>>> ? googletalk:conf+888 at conference.freeswitch.org
>>> ? pstn:+19193869900
>>> 
>>> ? 
> _________________________________________________________________________
>>> ? Professional FreeSWITCH Consulting Services:
>>> ? consulting at freeswitch.org
>>> ? http://www.freeswitchsolutions.com
>>> 
>>> ? 
>>> ? 
>>> 
>>> ? Official FreeSWITCH Sites
>>> ? http://www.freeswitch.org
>>> ? http://wiki.freeswitch.org
>>> ? http://www.cluecon.com
>>> 
>>> ? FreeSWITCH-users mailing list
>>> ? FreeSWITCH-users at lists.freeswitch.org
>>> ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> ? 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> ? http://www.freeswitch.org
>> 
>> 
>>  _________________________________________________________________________
>>  Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org
>>  http://www.freeswitchsolutions.com
>> 
>>  
>>  
>> 
>>  Official FreeSWITCH Sites
>>  http://www.freeswitch.org
>>  http://wiki.freeswitch.org
>>  http://www.cluecon.com
>> 
>>  FreeSWITCH-users mailing list
>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From msc at freeswitch.org  Fri Sep  7 02:35:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Sep 2012 15:35:03 -0700
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
Message-ID: <CAKzWOxVOqtNU6Y6VBR-QYgu2Fy6U3XGD1Xiw3yUkq-c2Wj2qCQ@mail.gmail.com>

No worries - it is indeed a total paradigm shift and things may seem
unusual. If you have a PBX registered with FreeSWITCH then from the FS
perspective it's a "user". Don't let the name "user" fool you - it's just a
label, and it's easier to write than "SIP registered endpoint."

So, if your trunk is registered then that means you've got an entry in
conf/directory/default/ that defines the "user". An example in there
already is 1000.xml, where the id="1000". So to route a call to that "user"
just do this:

<action application="bridge" data="user/1000"/>

That's it! FS does a lot of magic behind the scenes. So in the case of your
DID, you need to add a dialplan entry. I recommend making a copy of
conf/dialplan/default/00_inbound_did.xml and editing it to suit your needs.
Let's say that your DID is 8005551212 and that your PBX is registered as
user 1234. This dialplan entry would route an inbound DID call to your PBX:

<extension name="DID">
  <condition field="destination_number" expression="^\+?1?8005551212">
    <action application="bridge" data="user/1234"/>
  </condition>
</extension>

Note I added some regex magic to strip out optional leading + or 1.

Let us know how that works. Be sure to join IRC if you want to discuss it
in real time.
-MC

On Thu, Sep 6, 2012 at 12:09 PM, Matt Putnam <matt.putnam at lightspar.com>wrote:

> Sorry for the confusion I can really only equate it to asterisk as that?s
> what our current platform is. Essentially the problem is that if I have a
> PBX that has a trunk that is registered to freeswitch and a call is
> received for a DID  that is on that trunk I am currently unable to send
> that DID to the PBX. When the call is received on the PBX side it looks as
> if the call is for the trunk instead of the DID is should be for. So in the
> SIP invite instead of using lightspar1 at domain it would use 5551212 at domain.In asterisk this was simply accomplished by a dial statement of
> (SIP/5551212 at lightspar1) what I am looking for is that equivalent in
> freeswitch.****
>
> ** **
>
> Thanks,****
>
> Matt Putnam****
>
> matt.putnam at lightspar.com****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, September 06, 2012 10:48 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch****
>
> ** **
>
> I'm confused. I've read your message about three times, still don't
> completely understand.****
>
> Brian Foster
> Endigo Computer LLC****
>
> Sent from a mobile device.****
>
> On Sep 6, 2012 10:44 AM, "Matt Putnam" <matt.putnam at lightspar.com> wrote:*
> ***
>
> Thanks for the Response Mike I am able to get the trunk registered the
> real issue I guess is passing calls to that trunk. As an example I have
> freeswitch1 for customers which has a trunk lightspar1 registered to my
> gateway freeswitch box. When a call is placed to a DID that is associated
> to freeswitch1 I can see the call in the logs but the destination number is
> lightspar1 not the original DID that was called. Is there a way to set the
> from field to use the DID called instead of the trunk name so instead of
> lightspar1 at blah it would send the invite with NPANXXXXXX at blah?****
>
>  ****
>
> Thanks,****
>
> Matt Putnam****
>
> matt.putnam at lightspar.com****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, September 05, 2012 6:49 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch****
>
>  ****
>
> Hi Matt,
>
> Welcome to FreeSWITCH!
>
> If you want something to register with FreeSWITCH then simply add an entry
> in the user directory. Whether it's a "user" or a "trunk" really doesn't
> matter - it's just a SIP registration. Look in
> conf/directory/default/1000.xml (if using the example "vanilla"
> configuration) for a taste of what you need. Let us know if you have any
> other questions or join us in #freeswitch on irc.freenode.net.
>
> -MC (IRC:mercutioviz)****
>
> On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com>
> wrote:****
>
> This may have been asked before but my research hasn?t turned up any
> results. I am currently testing freeswitch as a replacement to our current
> asterisk gateways. I think I have the basics of ip based trunks down but I
> am running into an issue with trunks registering to the freeswitch box. In
> asterisk it was a fairly simple process of giving the trunk a username and
> password and setting it to dynamic to get it to register. What would be the
> equivalent in freeswitch that would allow me to have customer trunks
> register to the system? ****
>
>  ****
>
>  ****
>
> Thanks,****
>
> Matt Putnam****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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****
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12*
> ***
>
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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: 2012.0.2197 / Virus Database: 2437/5252 - Release Date: 09/06/12*
> ***
>
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lists at kavun.ch  Fri Sep  7 02:39:53 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 18:39:53 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
	<38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
	<1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <030CB8E7-4C92-4719-A74C-9F2B2620D4B4@kavun.ch>

Thanks Stanislav, your idea is good, but works only for incoming calls. I know we could trick the idea around to have it work for outbound calls as well, but it sounds to me as being too much of a workaround for something so trivial.

I'll play around some more and loop back my calls if I have to.

Thanks for this and have a good night,
Emrah
On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> Emrah, I think this particular use case should be quite easy to program in a Lua script.
> The incoming call is handed to the script, the script originates a call to the operator, 
> bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
> 
> 
> on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
> 
> 
> 
> 
> ----- Original Message -----
>> From: Emrah <lists at kavun.ch>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc: 
>> Sent: Thursday, September 6, 2012 5:57 PM
>> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
>> 
>> Hi Anthony,
>> 
>> The reason for the attended transfer to the conference call is to announce 
>> "manually" a new participant before releasing the transfer. E.g.: in 
>> the scenario where you have an operator transferring calling parties in a 
>> conference.
>> Which of the two alternatives would be the most CPU efficient? A loopback or a 
>> SIP call to the same domain? I wish I could give a hand in the code, but my 2 
>> cents would be to optimize this at some point in your roadmap.
>> 
>> Best,
>> Emrah
>> On Sep 4, 2012, at 1:17 PM, Anthony Minessale 
>> <anthony.minessale at gmail.com> wrote:
>> 
>>> Oh,
>>> 
>>> And you should try to avoid doing an attended transfer to
>>> one-legged-calls like conference or ivr, blind transfers work better
>>> for this because calls to apps are not bridged and the concept of
>>> transferring becomes confusing.  The other alternative is to bridge to
>>> the conference by looping the call over loopback or calling to the
>>> same box on sip so there is a true bridge.  But blind transferring is
>>> the best solution.
>>> 
>>> 
>>> On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> you are missing some of the sip and make sure its GIT HEAD (this
>>>> should be on jira)
>>>> 
>>>> sofia global siptrace on
>>>> 
>>>> 
>>>> 
>>>> On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
>>>>> Hi MC,
>>>>> 
>>>>> Thanks a bunch for your reply, sorry for the delay.
>>>>> Here are my logs:
>>>>> http://pastebin.freeswitch.org/19831
>>>>> 
>>>>> Any idea would be greatly appreciated.
>>>>> 
>>>>> Best,
>>>>> Emrah
>>>>> 
>>>>> 
>>>>> On Aug 28, 2012, at 10:56 AM, Michael Collins 
>> <msc at freeswitch.org> wrote:
>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> 
>>>>>> 
>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> --
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>> 
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>> 
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Sep  7 02:44:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Sep 2012 15:44:30 -0700
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <030CB8E7-4C92-4719-A74C-9F2B2620D4B4@kavun.ch>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
	<38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
	<1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
	<030CB8E7-4C92-4719-A74C-9F2B2620D4B4@kavun.ch>
Message-ID: <CAKzWOxXBbWg4=wTJVQvndAf_X9Xw8La__CXQ2Dnyx_cN2AZUQQ@mail.gmail.com>

Is the issue simply that you need to call into a conference and announce
the user before completing the transfer?
-MC

On Thu, Sep 6, 2012 at 3:39 PM, Emrah <lists at kavun.ch> wrote:

> Thanks Stanislav, your idea is good, but works only for incoming calls. I
> know we could trick the idea around to have it work for outbound calls as
> well, but it sounds to me as being too much of a workaround for something
> so trivial.
>
> I'll play around some more and loop back my calls if I have to.
>
> Thanks for this and have a good night,
> Emrah
> On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin at yahoo.com>
> wrote:
>
> > Emrah, I think this particular use case should be quite easy to program
> in a Lua script.
> > The incoming call is handed to the script, the script originates a call
> to the operator,
> > bridges them, then waits for operator's input, then hangs up the
> operator and bridges the A leg into the conference.
> >
> >
> > on the other side, why would anyone need a human operator in front of a
> conference bridge? 4 or 5 digits PIN usually works fine.
> >
> >
> >
> >
> > ----- Original Message -----
> >> From: Emrah <lists at kavun.ch>
> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> Cc:
> >> Sent: Thursday, September 6, 2012 5:57 PM
> >> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
> >>
> >> Hi Anthony,
> >>
> >> The reason for the attended transfer to the conference call is to
> announce
> >> "manually" a new participant before releasing the transfer. E.g.: in
> >> the scenario where you have an operator transferring calling parties in
> a
> >> conference.
> >> Which of the two alternatives would be the most CPU efficient? A
> loopback or a
> >> SIP call to the same domain? I wish I could give a hand in the code,
> but my 2
> >> cents would be to optimize this at some point in your roadmap.
> >>
> >> Best,
> >> Emrah
> >> On Sep 4, 2012, at 1:17 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>
> >>> Oh,
> >>>
> >>> And you should try to avoid doing an attended transfer to
> >>> one-legged-calls like conference or ivr, blind transfers work better
> >>> for this because calls to apps are not bridged and the concept of
> >>> transferring becomes confusing.  The other alternative is to bridge to
> >>> the conference by looping the call over loopback or calling to the
> >>> same box on sip so there is a true bridge.  But blind transferring is
> >>> the best solution.
> >>>
> >>>
> >>> On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>> you are missing some of the sip and make sure its GIT HEAD (this
> >>>> should be on jira)
> >>>>
> >>>> sofia global siptrace on
> >>>>
> >>>>
> >>>>
> >>>> On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
> >>>>> Hi MC,
> >>>>>
> >>>>> Thanks a bunch for your reply, sorry for the delay.
> >>>>> Here are my logs:
> >>>>> http://pastebin.freeswitch.org/19831
> >>>>>
> >>>>> Any idea would be greatly appreciated.
> >>>>>
> >>>>> Best,
> >>>>> Emrah
> >>>>>
> >>>>>
> >>>>> On Aug 28, 2012, at 10:56 AM, Michael Collins
> >> <msc at freeswitch.org> wrote:
> >>>>>
> >>>>>> 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
> >>>>>>
> >>>>>>
> >>>>>>
> >>
> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>
> >>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>>> IRC: irc.freenode.net #freeswitch
> >>>>
> >>>> FreeSWITCH Developer Conference
> >>>> sip:888 at conference.freeswitch.org
> >>>> googletalk:conf+888 at conference.freeswitch.org
> >>>> pstn:+19193869900
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> FreeSWITCH http://www.freeswitch.org/
> >>> ClueCon http://www.cluecon.com/
> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>
> >>> AIM: anthm
> >>> MSN:anthony_minessale at hotmail.com
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org
> >>> googletalk:conf+888 at conference.freeswitch.org
> >>> pstn:+19193869900
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ssinyagin at yahoo.com  Fri Sep  7 02:48:38 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 6 Sep 2012 15:48:38 -0700 (PDT)
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <030CB8E7-4C92-4719-A74C-9F2B2620D4B4@kavun.ch>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
	<38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
	<1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
	<030CB8E7-4C92-4719-A74C-9F2B2620D4B4@kavun.ch>
Message-ID: <1346971718.10919.YahooMailNeo@web39305.mail.mud.yahoo.com>

for outgoing calls, this could be a different Lua script, or the same script that handles both modes:

so, the conference moderator calls the destination number with some unique special prefix, this call gets into the script, then script dials out and connects B leg with the moderator, and after that waits for moderator's input to bridge the called person into a conference.

also if I'm not mistaken, I've seen somewhere an extension to the strandard freeswitch conference which allows to dialout and bring new people into the room. 

in any case, you can easily callout and bring people into the room from FreeSWITCH command line. This can also scripted in some way.






----- Original Message -----
> From: Emrah <lists at kavun.ch>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Friday, September 7, 2012 12:39 AM
> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
> 
>T hanks Stanislav, your idea is good, but works only for incoming calls. I know 
> we could trick the idea around to have it work for outbound calls as well, but 
> it sounds to me as being too much of a workaround for something so trivial.
> 
> I'll play around some more and loop back my calls if I have to.
> 
> Thanks for this and have a good night,
> Emrah
> On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> 
> wrote:
> 
>>  Emrah, I think this particular use case should be quite easy to program in 
> a Lua script.
>>  The incoming call is handed to the script, the script originates a call to 
> the operator, 
>>  bridges them, then waits for operator's input, then hangs up the 
> operator and bridges the A leg into the conference.
>> 
>> 
>>  on the other side, why would anyone need a human operator in front of a 
> conference bridge? 4 or 5 digits PIN usually works fine.
>> 
>> 
>> 
>> 
>>  ----- Original Message -----
>>>  From: Emrah <lists at kavun.ch>
>>>  To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>  Cc: 
>>>  Sent: Thursday, September 6, 2012 5:57 PM
>>>  Subject: Re: [Freeswitch-users] Attended transfer to a conference room
>>> 
>>>  Hi Anthony,
>>> 
>>>  The reason for the attended transfer to the conference call is to 
> announce 
>>>  "manually" a new participant before releasing the transfer. 
> E.g.: in 
>>>  the scenario where you have an operator transferring calling parties in 
> a 
>>>  conference.
>>>  Which of the two alternatives would be the most CPU efficient? A 
> loopback or a 
>>>  SIP call to the same domain? I wish I could give a hand in the code, 
> but my 2 
>>>  cents would be to optimize this at some point in your roadmap.
>>> 
>>>  Best,
>>>  Emrah
>>>  On Sep 4, 2012, at 1:17 PM, Anthony Minessale 
>>>  <anthony.minessale at gmail.com> wrote:
>>> 
>>>>  Oh,
>>>> 
>>>>  And you should try to avoid doing an attended transfer to
>>>>  one-legged-calls like conference or ivr, blind transfers work 
> better
>>>>  for this because calls to apps are not bridged and the concept of
>>>>  transferring becomes confusing.? The other alternative is to bridge 
> to
>>>>  the conference by looping the call over loopback or calling to the
>>>>  same box on sip so there is a true bridge.? But blind transferring 
> is
>>>>  the best solution.
>>>> 
>>>> 
>>>>  On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
>>>>  <anthony.minessale at gmail.com> wrote:
>>>>>  you are missing some of the sip and make sure its GIT HEAD 
> (this
>>>>>  should be on jira)
>>>>> 
>>>>>  sofia global siptrace on
>>>>> 
>>>>> 
>>>>> 
>>>>>  On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> 
> wrote:
>>>>>>  Hi MC,
>>>>>> 
>>>>>>  Thanks a bunch for your reply, sorry for the delay.
>>>>>>  Here are my logs:
>>>>>>  http://pastebin.freeswitch.org/19831
>>>>>> 
>>>>>>  Any idea would be greatly appreciated.
>>>>>> 
>>>>>>  Best,
>>>>>>  Emrah
>>>>>> 
>>>>>> 
>>>>>>  On Aug 28, 2012, at 10:56 AM, Michael Collins 
>>>  <msc at freeswitch.org> wrote:
>>>>>> 
>>>>>>>  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
>>>>>>> 
>>>>>>>  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
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>>  --
>>>>>>>  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
>>>>>>> 
>>>>>>>  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
>>>>>> 
>>>>>> 
>>>>>> 
>>> 
> _________________________________________________________________________
>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>  consulting at freeswitch.org
>>>>>>  http://www.freeswitchsolutions.com
>>>>>> 
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>>  Official FreeSWITCH Sites
>>>>>>  http://www.freeswitch.org
>>>>>>  http://wiki.freeswitch.org
>>>>>>  http://www.cluecon.com
>>>>>> 
>>>>>>  FreeSWITCH-users mailing list
>>>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>>  --
>>>>>  Anthony Minessale II
>>>>> 
>>>>>  FreeSWITCH http://www.freeswitch.org/
>>>>>  ClueCon http://www.cluecon.com/
>>>>>  Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> 
>>>>>  AIM: anthm
>>>>>  MSN:anthony_minessale at hotmail.com
>>>>>  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>  IRC: irc.freenode.net #freeswitch
>>>>> 
>>>>>  FreeSWITCH Developer Conference
>>>>>  sip:888 at conference.freeswitch.org
>>>>>  googletalk:conf+888 at conference.freeswitch.org
>>>>>  pstn:+19193869900
>>>> 
>>>> 
>>>> 
>>>>  -- 
>>>>  Anthony Minessale II
>>>> 
>>>>  FreeSWITCH http://www.freeswitch.org/
>>>>  ClueCon http://www.cluecon.com/
>>>>  Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>>  AIM: anthm
>>>>  MSN:anthony_minessale at hotmail.com
>>>>  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>  IRC: irc.freenode.net #freeswitch
>>>> 
>>>>  FreeSWITCH Developer Conference
>>>>  sip:888 at conference.freeswitch.org
>>>>  googletalk:conf+888 at conference.freeswitch.org
>>>>  pstn:+19193869900
>>>> 
>>>> 
> _________________________________________________________________________
>>>>  Professional FreeSWITCH Consulting Services:
>>>>  consulting at freeswitch.org
>>>>  http://www.freeswitchsolutions.com
>>>> 
>>>>  
>>>>  
>>>> 
>>>>  Official FreeSWITCH Sites
>>>>  http://www.freeswitch.org
>>>>  http://wiki.freeswitch.org
>>>>  http://www.cluecon.com
>>>> 
>>>>  FreeSWITCH-users mailing list
>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  http://www.freeswitch.org
>>> 
>>> 
>>> 
> _________________________________________________________________________
>>>  Professional FreeSWITCH Consulting Services:
>>>  consulting at freeswitch.org
>>>  http://www.freeswitchsolutions.com
>>> 
>>>  
>>>  
>>> 
>>>  Official FreeSWITCH Sites
>>>  http://www.freeswitch.org
>>>  http://wiki.freeswitch.org
>>>  http://www.cluecon.com
>>> 
>>>  FreeSWITCH-users mailing list
>>>  FreeSWITCH-users at lists.freeswitch.org
>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  http://www.freeswitch.org
>>> 
>> 
>>  _________________________________________________________________________
>>  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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>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From anthony.minessale at gmail.com  Fri Sep  7 03:36:53 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Sep 2012 18:36:53 -0500
Subject: [Freeswitch-users] att_xfer limitations
In-Reply-To: <015D0E1F-03A8-4452-B0F0-8657C273B5E4@kavun.ch>
References: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
	<CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>
	<D7E5118B-13F6-44F5-899E-B59225857F32@kavun.ch>
	<CABH8QX1B-4VSR7C-AoZCK3o3yyv27777JOYfEqgOud9BQVstxA@mail.gmail.com>
	<CAKbxfG8dexu-3yJEYG8qjcunAWi8wzsqR+mGjTRRiJgvHfzS1w@mail.gmail.com>
	<015D0E1F-03A8-4452-B0F0-8657C273B5E4@kavun.ch>
Message-ID: <CAKbxfG9kaCs9mF8uW4UkPnp1tNEiFmokAwT4sap5-v+7-UX+wQ@mail.gmail.com>

On Thu, Sep 6, 2012 at 3:09 PM, Emrah <lists at kavun.ch> wrote:

> Dear Anthony,
>
> You're message did not come across very pleasantly, but as it is often the
> case in writing, sometimes the receiver does not perceive things as they
> were meant originally. I'll disregard the judgmental portion of your email.
>
>
I can see you might have considered what I said judgmental however its
purely an observation.   Had you contended that we must design ships to
deal with the possibility of plummeting from the earth when they reached
the edge and I suggested you study the world geography it would not
be judgmental but rather an observation that you should learn more before
making a suggestion.  We do not mind criticism because feedback is the
cornerstone of open source software.  We just need to channel it to the
right place and in this case its not helpful.



> When you refer to the application as a whole, should I assume FreeSWITCH
> or ATT_XFER?
> When you mean thinking outside the box, am I to assume creating work
> arounds (e.g.: by using loopback calls?) I don't find that quite sexy to be
> honest.
>
>
I am referring to the application as a whole.  att_xfer is one small app
designed as a last resort for some specific situations.  Your suggestion to
engineer it differently is similar to saying you want your TV set to play
movies some other way than by connecting it to a source of video.  It
suggests you need to study it closer to understand properly.  The dialplan
is something that happens when a call goes to the ROUTING state and
collects a list of applications to execute.  Some of these applications can
execute more applications on top of each other such as running att_xfer
from digit action while already in the EXECUTE state during a call bridge.

Now in this case you want to dial another new channel and bridge to it then
subsequently either bridge it with the channel you are currently putting on
hold or end the call and go back to the original call.  So this concept of
needing a new channel is basically a given and is absolute.

So, you want to be able to make a call based on your dial-plan logic?
 Right, so, If this call lead to another box it would be a given, you call
it on sip or pstn etc, but since its on the same box its sort of a
catch-22.

Ok so there is a loopback channel which is basically allowing you to call
an extension on your own box in the context of a new channel so you can
properly transfer it etc.  The same thing can also be achieved by calling
the extension over SIP in a loop back to your own box either way the point
is you need to have another channel so you actually have something that can
continue on its own when you decide to bridge it to the other leg.  I can
assure you I had no intention of making it sexy when I made the loopback
module but I do often wish I never wrote it but then again it does solve a
particular set of problems.



> My previous message never demanded or even asked for an upgrade of a
> functionality. I stated my thoughts and encouraged the developers to
> consider my approach and see whether it was worth a revision. I obviously
> did not mean to attack your work in any way.
>
> I never assumed that to be the case.  Fret not.



> So here are my 2 cents:
> The att_xfer feature has been implemented for a certain reason. I've seen
> it working differently on Asterisk where the dialplan is used all the time.
> Based on that, and as a personal thought worth mentioning, I suggested a
> similar approach. Leaving the feature as it is, and based on the limited
> knowledge I currently have on FreeSWITCH, it makes it unusable in any of my
> use cases.
>
>
You are probably not aware but I personally implemented that original
feature in Asterisk "atxfer" in res_features and I can tell you that trying
to get it to work and how horrible it was to get working on the inside was
much more horrendous and as far from sexy as you can imagine even compared
to your distain for loopback channel.

Oh yeah, what we call loopback in Asterisk is called chan_local (you may
recall the Local/1234 type dial strings) and I can assure you that is the
exact method it uses to make it possible to dial right to extensions for
all the same rationale.


> I divorced my Asterisk boxes and made a commitment to FS. You've built
> something amazing and I will always try to be constructive where I can.
>
> Best to you Anthony, and again congrats for the super initiative you took
> by creating FS.
>
> Thank you.  I again encourage you to continue to study how things work so
you can get the most out of your usage of our software and feel free to
post any questions or comments you may have.




> Emrah
>
>
> On Sep 6, 2012, at 3:23 PM, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> > Emrah,
> >
> > I suggest you further study the application as a whole before making
> determinations on where improvements need to be made.
> > You may need to think a bit more "out of the box" if you want to be
> successful.
> >
> >
> >
> > On Thu, Sep 6, 2012 at 2:03 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
> wrote:
> > I do believe that there are some legit cases where att_xfer is an
> application that needs to be fully functional. The most obvious case is
> where analog phone are used with boards such as Sangoma, Khomp or Digium
> boards. It is in this instance arguable if that needs to be implemented as
> an application on the "core" so to speak on or the endpoint module.
> >
> > It is not a trivial task (at least beyond my capabilities) to make some
> extra things work on att_xfer such as call return.
> >
> > Nonetheless, Emrah, what you are asking is doable with the current
> implementation. Just use the loopback channel and make
> loopback_bowout=false. The downside in this case is that the loopback
> channel will be up during the entire duration of the call, but everything
> else will work.
> >
> > Also, this adds some extra complexity to CDR processing, but I haven't
> worked on that to really know how much complexity.
> >
> > I hope this works for you.
> >
> > Regards,
> > Jo?o Mesquita
> >
> >
> >
> >
> > On Thu, Sep 6, 2012 at 12:46 PM, Emrah <lists at kavun.ch> wrote:
> > Hi Michael,
> >
> > It was great to hear you as well, will try to be a regular. :)
> >
> > I use a Polycom and it does have all the capabilities I need, but I
> wanted to have the PBX side working for a couple of scenarios where I
> thought I might need it.
> > E.g.: pick up a call on your cellphone, walk into the office, att_xfer
> the call from your cell to your desk and continue the conversation. The
> reason of using att_xfer is to make sure to release the call when the line
> is properly established and prevent calls from landing in voicemail
> inadvertently.
> >
> > And on a final note, I never had any issue transferring calls from my
> SIP phones, but for some reason I find the Polycom way somewhat cumbersome
> and repelling.
> >
> > Best,
> > Emrah
> > On Sep 6, 2012, at 11:23 AM, Michael Collins <msc at freeswitch.org> wrote:
> >
> > > What kind of phone are you using? The reason I ask is that most desk
> phones have real transfer keys that do all of this. If you're stuck using
> something like x-lite then yeah, I can see the dilemma. In all honesty, the
> att_xfer app wasn't designed as the be all end all of call transfers. If
> you are in a scenario where your only option is to use att_xfer then I
> recommend solving that problem. In the long run it will be much better for
> you.
> > >
> > > -MC
> > >
> > > P.S. - It was nice having you on the conf call yesterday! Hope you can
> join more calls. :)
> > >
> > > On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:
> > > Hi guys,
> > >
> > > I just briefly tried the att_xfer app and it seems very limited to me.
> It seems to be either using a gateway or the db, but it never hits the
> dialplan at any point.
> > > I love the extra add-ons you guys built, especially the option to
> conference before finalizing the transfer, but I think the app should be
> revised at some point to support a more dialplan oriented approach.
> > > Now, when a call comes in, I can att_xfer to a SIP user, but can't
> call out a cellphone using my dialplan logic with my multiple outbound
> providers.
> > >
> > > I encourage you to give it a look and see if it's worth an upgrade.
> > >
> > > Best as always,
> > > 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
> > >
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
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From lists at kavun.ch  Fri Sep  7 03:44:27 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 19:44:27 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <CAKzWOxXBbWg4=wTJVQvndAf_X9Xw8La__CXQ2Dnyx_cN2AZUQQ@mail.gmail.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
	<38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
	<1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
	<030CB8E7-4C92-4719-A74C-9F2B2620D4B4@kavun.ch>
	<CAKzWOxXBbWg4=wTJVQvndAf_X9Xw8La__CXQ2Dnyx_cN2AZUQQ@mail.gmail.com>
Message-ID: <5CC45467-336E-4639-84A9-89B1A6838A10@kavun.ch>

Yes Sir, it pretty much sums up this use case. I also have phones that only support attended transfers and so it really is my only choice unless I use a PBX feature.

On Sep 6, 2012, at 6:44 PM, Michael Collins <msc at freeswitch.org> wrote:

> Is the issue simply that you need to call into a conference and announce the user before completing the transfer?
> -MC
> 
> On Thu, Sep 6, 2012 at 3:39 PM, Emrah <lists at kavun.ch> wrote:
> Thanks Stanislav, your idea is good, but works only for incoming calls. I know we could trick the idea around to have it work for outbound calls as well, but it sounds to me as being too much of a workaround for something so trivial.
> 
> I'll play around some more and loop back my calls if I have to.
> 
> Thanks for this and have a good night,
> Emrah
> On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> 
> > Emrah, I think this particular use case should be quite easy to program in a Lua script.
> > The incoming call is handed to the script, the script originates a call to the operator,
> > bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
> >
> >
> > on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
> >
> >
> >
> >
> > ----- Original Message -----
> >> From: Emrah <lists at kavun.ch>
> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> Cc:
> >> Sent: Thursday, September 6, 2012 5:57 PM
> >> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
> >>
> >> Hi Anthony,
> >>
> >> The reason for the attended transfer to the conference call is to announce
> >> "manually" a new participant before releasing the transfer. E.g.: in
> >> the scenario where you have an operator transferring calling parties in a
> >> conference.
> >> Which of the two alternatives would be the most CPU efficient? A loopback or a
> >> SIP call to the same domain? I wish I could give a hand in the code, but my 2
> >> cents would be to optimize this at some point in your roadmap.
> >>
> >> Best,
> >> Emrah
> >> On Sep 4, 2012, at 1:17 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>
> >>> Oh,
> >>>
> >>> And you should try to avoid doing an attended transfer to
> >>> one-legged-calls like conference or ivr, blind transfers work better
> >>> for this because calls to apps are not bridged and the concept of
> >>> transferring becomes confusing.  The other alternative is to bridge to
> >>> the conference by looping the call over loopback or calling to the
> >>> same box on sip so there is a true bridge.  But blind transferring is
> >>> the best solution.
> >>>
> >>>
> >>> On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>> you are missing some of the sip and make sure its GIT HEAD (this
> >>>> should be on jira)
> >>>>
> >>>> sofia global siptrace on
> >>>>
> >>>>
> >>>>
> >>>> On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> wrote:
> >>>>> Hi MC,
> >>>>>
> >>>>> Thanks a bunch for your reply, sorry for the delay.
> >>>>> Here are my logs:
> >>>>> http://pastebin.freeswitch.org/19831
> >>>>>
> >>>>> Any idea would be greatly appreciated.
> >>>>>
> >>>>> Best,
> >>>>> Emrah
> >>>>>
> >>>>>
> >>>>> On Aug 28, 2012, at 10:56 AM, Michael Collins
> >> <msc at freeswitch.org> wrote:
> >>>>>
> >>>>>> 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
> >>>>>>
> >>>>>>
> >>>>>>
> >> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>
> >> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>>> IRC: irc.freenode.net #freeswitch
> >>>>
> >>>> FreeSWITCH Developer Conference
> >>>> sip:888 at conference.freeswitch.org
> >>>> googletalk:conf+888 at conference.freeswitch.org
> >>>> pstn:+19193869900
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> FreeSWITCH http://www.freeswitch.org/
> >>> ClueCon http://www.cluecon.com/
> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>
> >>> AIM: anthm
> >>> MSN:anthony_minessale at hotmail.com
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org
> >>> googletalk:conf+888 at conference.freeswitch.org
> >>> pstn:+19193869900
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fs-list at communicatefreely.net  Fri Sep  7 03:45:36 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 06 Sep 2012 19:45:36 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
Message-ID: <504935A0.9070106@communicatefreely.net>

Hi Emrah,

I have been using Aastra phones extensively.  If you stay with the 5xi 
series, you should get good sound quality.  The 6739i is really nice, 
but touch screen, so that's out.

There are a few different ways to provision them - the simplest way is 
for them to download a config file (ours are rendered on the fly with 
PHP).  You can also use the Web UI, or send the phone XML commands.  If 
you make changes in the web UI or push config via XML, the changes take 
place right away on the newer firmwares.  This is great for 
experimenting, as you can change SIP settings around without having to 
reboot every time.  You can also interface the phones to other systems 
using the XML interface, or via some proprietary software.  There is a 
third party app that lets you click to dial from your computer, and you 
get on-screen notifications of incoming calls.

Some versions of firmware are buggy, but they are getting better.  I'm 
waiting for IPv6 compatible firmware, but I'm told that it's in the 
works and they put me on the beta test list.

Hope that's helpful.

-Tim

Emrah wrote:
> Hi all,
>
> I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.
>
> 1. It's touch screen and I'm blind.
> 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.
>
> I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.
>
> What can you recommend that is stable, versatile, open and good quality? 
>
> Cheers,
> 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
>   



From lists at kavun.ch  Fri Sep  7 03:57:26 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 19:57:26 -0400
Subject: [Freeswitch-users] att_xfer limitations
In-Reply-To: <CAKbxfG9kaCs9mF8uW4UkPnp1tNEiFmokAwT4sap5-v+7-UX+wQ@mail.gmail.com>
References: <A33C5577-0314-4D6B-B49F-C1478A82DE7D@kavun.ch>
	<CAKzWOxV_o6scZXp=wRdDZ7c-V0AJbRr7DC3GJqBCKdPM8XAP2Q@mail.gmail.com>
	<D7E5118B-13F6-44F5-899E-B59225857F32@kavun.ch>
	<CABH8QX1B-4VSR7C-AoZCK3o3yyv27777JOYfEqgOud9BQVstxA@mail.gmail.com>
	<CAKbxfG8dexu-3yJEYG8qjcunAWi8wzsqR+mGjTRRiJgvHfzS1w@mail.gmail.com>
	<015D0E1F-03A8-4452-B0F0-8657C273B5E4@kavun.ch>
	<CAKbxfG9kaCs9mF8uW4UkPnp1tNEiFmokAwT4sap5-v+7-UX+wQ@mail.gmail.com>
Message-ID: <5EA8CD0F-D42B-43C4-8D7E-070A935D9D55@kavun.ch>

Got it, and I appreciate that you took the time to get back to me. I agree with all your points and things make more sense now.

Will keep on working my teeth on my FS box.

You got a beer from me next time you swing by NYC!

Cheers,
Emrah

On Sep 6, 2012, at 7:36 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> 
> 
> 
> On Thu, Sep 6, 2012 at 3:09 PM, Emrah <lists at kavun.ch> wrote:
> Dear Anthony,
> 
> You're message did not come across very pleasantly, but as it is often the case in writing, sometimes the receiver does not perceive things as they were meant originally. I'll disregard the judgmental portion of your email.
> 
> 
> I can see you might have considered what I said judgmental however its purely an observation.   Had you contended that we must design ships to deal with the possibility of plummeting from the earth when they reached the edge and I suggested you study the world geography it would not be judgmental but rather an observation that you should learn more before making a suggestion.  We do not mind criticism because feedback is the cornerstone of open source software.  We just need to channel it to the right place and in this case its not helpful.
> 
>  
> When you refer to the application as a whole, should I assume FreeSWITCH or ATT_XFER?
> When you mean thinking outside the box, am I to assume creating work arounds (e.g.: by using loopback calls?) I don't find that quite sexy to be honest.
> 
> 
> I am referring to the application as a whole.  att_xfer is one small app designed as a last resort for some specific situations.  Your suggestion to engineer it differently is similar to saying you want your TV set to play movies some other way than by connecting it to a source of video.  It suggests you need to study it closer to understand properly.  The dialplan is something that happens when a call goes to the ROUTING state and collects a list of applications to execute.  Some of these applications can execute more applications on top of each other such as running att_xfer from digit action while already in the EXECUTE state during a call bridge.  
> 
> Now in this case you want to dial another new channel and bridge to it then subsequently either bridge it with the channel you are currently putting on hold or end the call and go back to the original call.  So this concept of needing a new channel is basically a given and is absolute.  
> 
> So, you want to be able to make a call based on your dial-plan logic?  Right, so, If this call lead to another box it would be a given, you call it on sip or pstn etc, but since its on the same box its sort of a catch-22.  
> 
> Ok so there is a loopback channel which is basically allowing you to call an extension on your own box in the context of a new channel so you can properly transfer it etc.  The same thing can also be achieved by calling the extension over SIP in a loop back to your own box either way the point is you need to have another channel so you actually have something that can continue on its own when you decide to bridge it to the other leg.  I can assure you I had no intention of making it sexy when I made the loopback module but I do often wish I never wrote it but then again it does solve a particular set of problems.
> 
>  
> My previous message never demanded or even asked for an upgrade of a functionality. I stated my thoughts and encouraged the developers to consider my approach and see whether it was worth a revision. I obviously did not mean to attack your work in any way.
> 
> I never assumed that to be the case.  Fret not.
> 
>  
> So here are my 2 cents:
> The att_xfer feature has been implemented for a certain reason. I've seen it working differently on Asterisk where the dialplan is used all the time. Based on that, and as a personal thought worth mentioning, I suggested a similar approach. Leaving the feature as it is, and based on the limited knowledge I currently have on FreeSWITCH, it makes it unusable in any of my use cases.
> 
> 
> You are probably not aware but I personally implemented that original feature in Asterisk "atxfer" in res_features and I can tell you that trying to get it to work and how horrible it was to get working on the inside was much more horrendous and as far from sexy as you can imagine even compared to your distain for loopback channel. 
> 
> Oh yeah, what we call loopback in Asterisk is called chan_local (you may recall the Local/1234 type dial strings) and I can assure you that is the exact method it uses to make it possible to dial right to extensions for all the same rationale.
>  
> I divorced my Asterisk boxes and made a commitment to FS. You've built something amazing and I will always try to be constructive where I can.
> 
> Best to you Anthony, and again congrats for the super initiative you took by creating FS.
> 
> Thank you.  I again encourage you to continue to study how things work so you can get the most out of your usage of our software and feel free to post any questions or comments you may have.
> 
> 
>  
> Emrah
> 
> 
> On Sep 6, 2012, at 3:23 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
> > Emrah,
> >
> > I suggest you further study the application as a whole before making determinations on where improvements need to be made.
> > You may need to think a bit more "out of the box" if you want to be successful.
> >
> >
> >
> > On Thu, Sep 6, 2012 at 2:03 PM, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
> > I do believe that there are some legit cases where att_xfer is an application that needs to be fully functional. The most obvious case is where analog phone are used with boards such as Sangoma, Khomp or Digium boards. It is in this instance arguable if that needs to be implemented as an application on the "core" so to speak on or the endpoint module.
> >
> > It is not a trivial task (at least beyond my capabilities) to make some extra things work on att_xfer such as call return.
> >
> > Nonetheless, Emrah, what you are asking is doable with the current implementation. Just use the loopback channel and make loopback_bowout=false. The downside in this case is that the loopback channel will be up during the entire duration of the call, but everything else will work.
> >
> > Also, this adds some extra complexity to CDR processing, but I haven't worked on that to really know how much complexity.
> >
> > I hope this works for you.
> >
> > Regards,
> > Jo?o Mesquita
> >
> >
> >
> >
> > On Thu, Sep 6, 2012 at 12:46 PM, Emrah <lists at kavun.ch> wrote:
> > Hi Michael,
> >
> > It was great to hear you as well, will try to be a regular. :)
> >
> > I use a Polycom and it does have all the capabilities I need, but I wanted to have the PBX side working for a couple of scenarios where I thought I might need it.
> > E.g.: pick up a call on your cellphone, walk into the office, att_xfer the call from your cell to your desk and continue the conversation. The reason of using att_xfer is to make sure to release the call when the line is properly established and prevent calls from landing in voicemail inadvertently.
> >
> > And on a final note, I never had any issue transferring calls from my SIP phones, but for some reason I find the Polycom way somewhat cumbersome and repelling.
> >
> > Best,
> > Emrah
> > On Sep 6, 2012, at 11:23 AM, Michael Collins <msc at freeswitch.org> wrote:
> >
> > > What kind of phone are you using? The reason I ask is that most desk phones have real transfer keys that do all of this. If you're stuck using something like x-lite then yeah, I can see the dilemma. In all honesty, the att_xfer app wasn't designed as the be all end all of call transfers. If you are in a scenario where your only option is to use att_xfer then I recommend solving that problem. In the long run it will be much better for you.
> > >
> > > -MC
> > >
> > > P.S. - It was nice having you on the conf call yesterday! Hope you can join more calls. :)
> > >
> > > On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:
> > > Hi guys,
> > >
> > > I just briefly tried the att_xfer app and it seems very limited to me. It seems to be either using a gateway or the db, but it never hits the dialplan at any point.
> > > I love the extra add-ons you guys built, especially the option to conference before finalizing the transfer, but I think the app should be revised at some point to support a more dialplan oriented approach.
> > > Now, when a call comes in, I can att_xfer to a SIP user, but can't call out a cellphone using my dialplan logic with my multiple outbound providers.
> > >
> > > I encourage you to give it a look and see if it's worth an upgrade.
> > >
> > > Best as always,
> > > 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
> > >
> > >
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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



From lists at kavun.ch  Fri Sep  7 04:02:29 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Sep 2012 20:02:29 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <504935A0.9070106@communicatefreely.net>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
Message-ID: <8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>

That's great Tim, thanks! I'll check them out.
I purchased 3 Aastra dect phones for my apt (some clone of Snom M3) and was super disappointed by the quality? The speakerphone is simply unusable.

Regards,
Emrah
On Sep 6, 2012, at 7:45 PM, "Tim St. Pierre" <fs-list at communicatefreely.net> wrote:

> Hi Emrah,
> 
> I have been using Aastra phones extensively.  If you stay with the 5xi 
> series, you should get good sound quality.  The 6739i is really nice, 
> but touch screen, so that's out.
> 
> There are a few different ways to provision them - the simplest way is 
> for them to download a config file (ours are rendered on the fly with 
> PHP).  You can also use the Web UI, or send the phone XML commands.  If 
> you make changes in the web UI or push config via XML, the changes take 
> place right away on the newer firmwares.  This is great for 
> experimenting, as you can change SIP settings around without having to 
> reboot every time.  You can also interface the phones to other systems 
> using the XML interface, or via some proprietary software.  There is a 
> third party app that lets you click to dial from your computer, and you 
> get on-screen notifications of incoming calls.
> 
> Some versions of firmware are buggy, but they are getting better.  I'm 
> waiting for IPv6 compatible firmware, but I'm told that it's in the 
> works and they put me on the beta test list.
> 
> Hope that's helpful.
> 
> -Tim
> 
> Emrah wrote:
>> Hi all,
>> 
>> I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.
>> 
>> 1. It's touch screen and I'm blind.
>> 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
>> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.
>> 
>> I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.
>> 
>> What can you recommend that is stable, versatile, open and good quality? 
>> 
>> Cheers,
>> 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
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fs-list at communicatefreely.net  Fri Sep  7 04:04:24 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 06 Sep 2012 20:04:24 -0400
Subject: [Freeswitch-users] Phone that does IPv6 properly?
Message-ID: <50493A08.5020909@communicatefreely.net>

Hello,

I'm excited about moving to IPv6, and not having to deal with NAT ever 
again.

We have IPv6 address space and transit, and FS does IPv6, so now I need 
some phones.

I'm told that there will be IPv6 firmware coming for my 400+ Aastra 
phones, but that might be a while.  Are some IP phones out there today 
that have good IPv6 support?  Specifically, it has to do the usual 
network protocols (SLAC, router solicitation, etc.)  DHCP6 support would 
be nice.  It has to be able to look up an SRV record, and then get AAAA 
records for the SIP proxies referenced.  Everything is native IPv6, so 
it isn't really important that it can do tunnels or anything special.  
I'm not even sure that I need IPv4 on it at all, since it will be 
deployed in an IPv6 only environment.

G.722 audio would be nice too.

Thanks for any suggestions!

-Tim


From fs-list at communicatefreely.net  Fri Sep  7 04:39:23 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 06 Sep 2012 20:39:23 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
Message-ID: <5049423B.9070401@communicatefreely.net>

Yes, those were terrible.  We sold four of them, and have since replaced 
them all.

I think they were part of some acquisition that Aastra made, and they 
never really matured.  I wouldn't judge Aastra by that phone.

-Tim

Emrah wrote:
> That's great Tim, thanks! I'll check them out.
> I purchased 3 Aastra dect phones for my apt (some clone of Snom M3) and was super disappointed by the quality? The speakerphone is simply unusable.
>
> Regards,
> Emrah
> On Sep 6, 2012, at 7:45 PM, "Tim St. Pierre" <fs-list at communicatefreely.net> wrote:
>
>   
>> Hi Emrah,
>>
>> I have been using Aastra phones extensively.  If you stay with the 5xi 
>> series, you should get good sound quality.  The 6739i is really nice, 
>> but touch screen, so that's out.
>>
>> There are a few different ways to provision them - the simplest way is 
>> for them to download a config file (ours are rendered on the fly with 
>> PHP).  You can also use the Web UI, or send the phone XML commands.  If 
>> you make changes in the web UI or push config via XML, the changes take 
>> place right away on the newer firmwares.  This is great for 
>> experimenting, as you can change SIP settings around without having to 
>> reboot every time.  You can also interface the phones to other systems 
>> using the XML interface, or via some proprietary software.  There is a 
>> third party app that lets you click to dial from your computer, and you 
>> get on-screen notifications of incoming calls.
>>
>> Some versions of firmware are buggy, but they are getting better.  I'm 
>> waiting for IPv6 compatible firmware, but I'm told that it's in the 
>> works and they put me on the beta test list.
>>
>> Hope that's helpful.
>>
>> -Tim
>>
>> Emrah wrote:
>>     
>>> Hi all,
>>>
>>> I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.
>>>
>>> 1. It's touch screen and I'm blind.
>>> 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
>>> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.
>>>
>>> I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.
>>>
>>> What can you recommend that is stable, versatile, open and good quality? 
>>>
>>> Cheers,
>>> 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
>>>
>>>       
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Y


From fs-list at communicatefreely.net  Fri Sep  7 04:43:27 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 06 Sep 2012 20:43:27 -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: <5049432F.3090406@communicatefreely.net>

Hi Alex,

I see that this is an older thread, but I now have native IPv6 
connectivity set up at our datacenter, tunneled IPv6 at many customer 
sites with native on the way.

I agree that it gets rid of all the NAT issues, so I'm hoping to try it 
out very soon.  For larger installations, I'm insisting that native IPv6 
be available within the next few months.

I'm having a hard time finding phones that can do IPv6 though.

Any suggestions?



Alex 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
>   



From lists at kavun.ch  Fri Sep  7 08:45:42 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 7 Sep 2012 00:45:42 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <1346971718.10919.YahooMailNeo@web39305.mail.mud.yahoo.com>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
	<CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>
	<152C17A2-68AA-4EC5-AD2E-2529925A17D0@kavun.ch>
	<CAKbxfG-V-w3gzEuUprLM3qt5vpUe_Y7v4X0H-g5QU_HqJsgKNw@mail.gmail.com>
	<CAKbxfG9bYRfWj3Ns6XtyNZ9zq-o1UJtPw11NHZaDM5XjeMJjjw@mail.gmail.com>
	<38779F6D-256D-455D-952C-92300F12C9BF@kavun.ch>
	<1346969950.71314.YahooMailNeo@web39302.mail.mud.yahoo.com>
	<030CB8E7-4C92-4719-A74C-9F2B2620D4B4@kavun.ch>
	<1346971718.10919.YahooMailNeo@web39305.mail.mud.yahoo.com>
Message-ID: <C6B02C46-E813-494A-B0AA-F12E3DD38C82@kavun.ch>

I now realize that even blind transfers fail. There must be something wrong in my config. Will post the logs tomorrow.

On Sep 6, 2012, at 6:48 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> for outgoing calls, this could be a different Lua script, or the same script that handles both modes:
> 
> so, the conference moderator calls the destination number with some unique special prefix, this call gets into the script, then script dials out and connects B leg with the moderator, and after that waits for moderator's input to bridge the called person into a conference.
> 
> also if I'm not mistaken, I've seen somewhere an extension to the strandard freeswitch conference which allows to dialout and bring new people into the room. 
> 
> in any case, you can easily callout and bring people into the room from FreeSWITCH command line. This can also scripted in some way.
> 
> 
> 
> 
> 
> 
> ----- Original Message -----
>> From: Emrah <lists at kavun.ch>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc: 
>> Sent: Friday, September 7, 2012 12:39 AM
>> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
>> 
>> T hanks Stanislav, your idea is good, but works only for incoming calls. I know 
>> we could trick the idea around to have it work for outbound calls as well, but 
>> it sounds to me as being too much of a workaround for something so trivial.
>> 
>> I'll play around some more and loop back my calls if I have to.
>> 
>> Thanks for this and have a good night,
>> Emrah
>> On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> 
>> wrote:
>> 
>>> Emrah, I think this particular use case should be quite easy to program in 
>> a Lua script.
>>> The incoming call is handed to the script, the script originates a call to 
>> the operator, 
>>> bridges them, then waits for operator's input, then hangs up the 
>> operator and bridges the A leg into the conference.
>>> 
>>> 
>>> on the other side, why would anyone need a human operator in front of a 
>> conference bridge? 4 or 5 digits PIN usually works fine.
>>> 
>>> 
>>> 
>>> 
>>> ----- Original Message -----
>>>> From: Emrah <lists at kavun.ch>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Cc: 
>>>> Sent: Thursday, September 6, 2012 5:57 PM
>>>> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
>>>> 
>>>> Hi Anthony,
>>>> 
>>>> The reason for the attended transfer to the conference call is to 
>> announce 
>>>> "manually" a new participant before releasing the transfer. 
>> E.g.: in 
>>>> the scenario where you have an operator transferring calling parties in 
>> a 
>>>> conference.
>>>> Which of the two alternatives would be the most CPU efficient? A 
>> loopback or a 
>>>> SIP call to the same domain? I wish I could give a hand in the code, 
>> but my 2 
>>>> cents would be to optimize this at some point in your roadmap.
>>>> 
>>>> Best,
>>>> Emrah
>>>> On Sep 4, 2012, at 1:17 PM, Anthony Minessale 
>>>> <anthony.minessale at gmail.com> wrote:
>>>> 
>>>>> Oh,
>>>>> 
>>>>> And you should try to avoid doing an attended transfer to
>>>>> one-legged-calls like conference or ivr, blind transfers work 
>> better
>>>>> for this because calls to apps are not bridged and the concept of
>>>>> transferring becomes confusing.  The other alternative is to bridge 
>> to
>>>>> the conference by looping the call over loopback or calling to the
>>>>> same box on sip so there is a true bridge.  But blind transferring 
>> is
>>>>> the best solution.
>>>>> 
>>>>> 
>>>>> On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>> you are missing some of the sip and make sure its GIT HEAD 
>> (this
>>>>>> should be on jira)
>>>>>> 
>>>>>> sofia global siptrace on
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists at kavun.ch> 
>> wrote:
>>>>>>> Hi MC,
>>>>>>> 
>>>>>>> Thanks a bunch for your reply, sorry for the delay.
>>>>>>> Here are my logs:
>>>>>>> http://pastebin.freeswitch.org/19831
>>>>>>> 
>>>>>>> Any idea would be greatly appreciated.
>>>>>>> 
>>>>>>> Best,
>>>>>>> Emrah
>>>>>>> 
>>>>>>> 
>>>>>>> On Aug 28, 2012, at 10:56 AM, Michael Collins 
>>>> <msc at freeswitch.org> wrote:
>>>>>>> 
>>>>>>>> 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
>>>>>>>> 
>>>>>>>> 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
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> --
>>>>>>>> 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
>>>>>>>> 
>>>>>>>> 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
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>> 
>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>> 
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>> 
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>> 
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>> 
>>>>> 
>>>>> 
>>>>> -- 
>>>>> Anthony Minessale II
>>>>> 
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> 
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>> 
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>> 
>>>>> 
>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From embedded65 at gmail.com  Fri Sep  7 08:52:10 2012
From: embedded65 at gmail.com (Stephan Swart)
Date: Fri, 7 Sep 2012 06:52:10 +0200
Subject: [Freeswitch-users] More information on FreeSwitch
Message-ID: <CC20783F-329B-4377-914E-CE4D640529C6@gmail.com>

Hi

Obviously i am new around here
I am looking for more documentation on FreeSwitch

I am aware of the Wiki

I do have the cookbook

What else is around

I need to design a large voice deployment with Class 4 and Class 5 switches.

I need to know more detail and what is out there that can manage these devices in an easier manner than editing files. Administrators can be dangerous as well sometimes.

Need some proper stability etc.


Tanx
Stephan



From tichsig at gmail.com  Fri Sep  7 10:00:51 2012
From: tichsig at gmail.com (Tichafara Sigauke)
Date: Fri, 7 Sep 2012 08:00:51 +0200
Subject: [Freeswitch-users] gtalk rings once and cuts
Message-ID: <CAJSEwzL5E5i89O8zxNNRWLQe1YwVxfYUetDr8=SfSo=zT4oijA@mail.gmail.com>

I have managed to setup mod_dingaling on my Freeswitch, I would like to
route inbound calls from gtalk to my IVR running on Freeswitch but calls
cut after one ring.
My setup looks like:
gtalk client A-------------------------------->gtalk
Client at Freeswitch----------------->IVR on the same Freeswitch
I have attached the fs-cli debug fs_cli_debug.txt and profile setup
client_profile.txt, I would appreciate any help.

Tich
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freeswitch at internal> 
freeswitch at internal> 
2012-08-20 13:16:51.223514 [INFO] libdingaling.c:1743 SecRECV:
-------------------------------------------------------------------------------
<iq type="set" to="freedomfone01 at gmail.com/gtalkB846928D" id="EEAA9C5E22D149CE" from="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A">
  <jin:jingle action="session-initiate" sid="c403880385" initiator="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gtalkB846928D" xmlns:jin="urn:xmpp:jingle:1">
    <jin:content name="audio" creator="initiator">
      <rtp:description media="audio" ssrc="2747399142" xmlns:rtp="urn:xmpp:jingle:apps:rtp:1">
        <rtp:payload-type id="103" name="ISAC" clockrate="16000"></rtp:payload-type>
        <rtp:payload-type id="104" name="ISAC" clockrate="32000"></rtp:payload-type>
        <rtp:payload-type id="107" name="speex" clockrate="16000">
          <rtp:parameter name="bitrate" value="22000"></rtp:parameter>
        </rtp:payload-type>
        <rtp:payload-type id="9" name="G722" clockrate="16000">
          <rtp:parameter name="bitrate" value="64000"></rtp:parameter>
        </rtp:payload-type>
        <rtp:payload-type id="102" name="ILBC" clockrate="8000">
          <rtp:parameter name="bitrate" value="13300"></rtp:parameter>
        </rtp:payload-type>
        <rtp:payload-type id="108" name="speex" clockrate="8000"></rtp:payload-type>
      </rtp:description>
    </jin:content>
  </jin:jingle>
</iq>

2012-08-20 13:16:51.223514 [INFO] libdingaling.c:1743 SecRECV:
-------------------------------------------------------------------------------

2012-08-20 13:16:51.223514 [INFO] libdingaling.c:1743 SecRECV:
-------------------------------------------------------------------------------
<pho:description xmlns:pho="http://www.google.com/session/phone">
  <pho:payload-type id="103" name="ISAC" clockrate="16000"></pho:payload-type>
  <pho:payload-type id="104" name="ISAC" clockrate="32000"></pho:payload-type>
  <pho:payload-type id="107" name="speex" bitrate="22000" clockrate="16000"></pho:payload-type>
  <pho:payload-type id="9" name="G722" bitrate="64000" clockrate="16000"></pho:payload-type>
  <pho:payload-type id="102" name="ILBC" bitrate="13300" clockrate="8000"></pho:payload-type>
  <pho:payload-type id="108" name="speex" bitrate="11000" clockrate="8000"></pho:payload-type>
  <pho:payload-type id="0" name="PCMU" bitrate="64000" clockrate="8000"></pho:payload-type>
  <pho:payload-type id="8" name="PCMA" bitrate="64000" clockrate="8000"></pho:payload-type>
  <pho:payload-type id="127" name="red" clockrate="8000"></pho:payload-type>
  <pho:payload-type id="126" name="telephone-event" clockrate="8000"></pho:payload-type>
  <pho:rtcp-mux></pho:rtcp-mux>
  <pho:src-id>2747399142</pho:src-id>
</pho:description>

2012-08-20 13:16:51.223514 [INFO] libdingaling.c:1743 SecRECV:
-------------------------------------------------------------------------------
<rtp:crypto crypto-suite="AES_CM_128_HMAC_SHA1_80" key-params="inline:f9+THf/j6IE2L2No/fSufRsO3Kn1KGu/xzrW94He" tag="1"></rtp:crypto>

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:363 Created Session c403880385

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:391 Message for Session c403880385

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [ISAC] id='103'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [ISAC] id='104'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [speex] id='107'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [G722] id='9'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [ILBC] id='102'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [speex] id='108'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [PCMU] id='0'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [PCMA] id='8'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [red] id='127'

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:462 Add Payload [telephone-event] id='126'

2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:4204 Creating an identity for c403880385 2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A <2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A> 5555
2012-08-20 13:16:51.223514 [NOTICE] switch_channel.c:930 New Channel dingaling/5555 [b9e700dd-4124-41da-9a7e-46235f7482e5]
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:4232 Creating a session for c403880385
2012-08-20 13:16:51.223514 [NOTICE] switch_channel.c:928 Rename Channel dingaling/5555->DingaLing/new [b9e700dd-4124-41da-9a7e-46235f7482e5]
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:4236 (DingaLing/new) State Change CS_NEW -> CS_INIT
2012-08-20 13:16:51.223514 [DEBUG] switch_core_session.c:1229 Send signal DingaLing/new [BREAK]
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:2019 DingaLing/new CHANNEL KILL
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3696 10 payloads
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3698 Available Payload ISAC 103
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3711 compare ISAC 103/16000 to PCMU 0/8000
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3698 Available Payload ISAC 104
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3711 compare ISAC 104/32000 to PCMU 0/8000
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3698 Available Payload speex 107
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3711 compare speex 107/16000 to PCMU 0/8000
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3698 Available Payload G722 9
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3711 compare G722 9/16000 to PCMU 0/8000
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3698 Available Payload ILBC 102
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3711 compare ILBC 102/8000 to PCMU 0/8000
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3698 Available Payload speex 108
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3711 compare speex 108/8000 to PCMU 0/8000
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3698 Available Payload PCMU 0
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3711 compare PCMU 0/8000 to PCMU 0/8000
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:3723 Choosing rtp Payload index 0 PCMU 0
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:1670 Don't have video codec.
2012-08-20 13:16:51.223514 [DEBUG] mod_dingaling.c:1681 Send Describe [PCMU at 8000]
2012-08-20 13:16:51.223514 [NOTICE] libdingaling.c:1745 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A" from="freedomfone01 at gmail.com/gtalkB846928D" id="EEAA9C5E22D149CE">
  <query></query>
</iq>

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:16:51.223514 [CRIT] libdingaling.c:1847 Sending packet 312 (2 left)

2012-08-20 13:16:51.223514 [NOTICE] libdingaling.c:1745 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="freedomfone01 at gmail.com/gtalkB846928D" to="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A" type="set" id="312">
  <ses:session xmlns:ses="http://www.google.com/session" type="accept" id="c403880385" initiator="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gtalkB846928D">
    <pho:description xmlns:pho="http://www.google.com/session/phone">
      <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type>
    </pho:description>
  </ses:session>
</iq>

2012-08-20 13:16:51.223514 [DEBUG] switch_core_state_machine.c:385 (DingaLing/new) Running State Change CS_INIT
2012-08-20 13:16:51.223514 [DEBUG] switch_core_state_machine.c:424 (DingaLing/new) State INIT
2012-08-20 13:16:51.223514 [NOTICE] mod_dingaling.c:1748 Ring-Ready DingaLing/new!
2012-08-20 13:16:52.323477 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:16:53.423481 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:16:54.523557 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:16:55.623496 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:16:56.723481 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:16:56.723481 [CRIT] libdingaling.c:1847 Sending packet 312 (1 left)

2012-08-20 13:16:56.723481 [NOTICE] libdingaling.c:1745 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="freedomfone01 at gmail.com/gtalkB846928D" to="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A" type="set" id="312">
  <ses:session xmlns:ses="http://www.google.com/session" type="accept" id="c403880385" initiator="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gtalkB846928D">
    <pho:description xmlns:pho="http://www.google.com/session/phone">
      <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type>
    </pho:description>
  </ses:session>
</iq>

2012-08-20 13:16:57.143480 [INFO] libdingaling.c:1743 SecRECV:
-------------------------------------------------------------------------------
<iq from="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A" to="freedomfone01 at gmail.com/gtalkB846928D" type="error" id="312">
  <ses:session type="accept" id="c403880385" initiator="1erajjvau3can276eqgtt6ggk8 at public.talk.google.com/gmail.3AF62E0A" xmlns:ses="http://www.google.com/session">
    <pho:description xmlns:pho="http://www.google.com/session/phone">
      <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type>
    </pho:description>
  </ses:session>
  <error code="405" type="cancel">
    <not-allowed xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></not-allowed>
  </error>
</iq>

2012-08-20 13:16:57.143480 [CRIT] libdingaling.c:1305 Cancel packet 312

2012-08-20 13:16:57.143480 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:16:57.143480 [CRIT] libdingaling.c:1856 Discarding packet 312

2012-08-20 13:17:01.243460 [DEBUG] mod_dingaling.c:1496 Stun Lookup Local 192.168.122.121:24294
2012-08-20 13:17:02.403478 [INFO] mod_dingaling.c:1506 Stun Success 196.44.184.138:24294
2012-08-20 13:17:02.423533 [DEBUG] mod_dingaling.c:1520 Send rtp Candidate 196.44.184.138:24294 [LHrfCI0Jp6gNld6x]
2012-08-20 13:17:02.423533 [DEBUG] mod_dingaling.c:1496 Stun Lookup Local 192.168.122.121:24295
2012-08-20 13:17:02.703468 [INFO] mod_dingaling.c:1506 Stun Success 196.44.184.138:24295
2012-08-20 13:17:02.703468 [DEBUG] mod_dingaling.c:1520 Send rtcp Candidate 196.44.184.138:24295 [lhHB2Wam6U6KOPru]
2012-08-20 13:17:02.703468 [DEBUG] mod_dingaling.c:1496 Stun Lookup Local 192.168.122.121:24296
2012-08-20 13:17:03.003441 [INFO] mod_dingaling.c:1506 Stun Success 196.44.184.138:24296
2012-08-20 13:17:03.003441 [DEBUG] mod_dingaling.c:1520 Send video_rtp Candidate 196.44.184.138:24296 [3bukkQnujVKIexCk]
2012-08-20 13:17:03.003441 [DEBUG] mod_dingaling.c:1496 Stun Lookup Local 192.168.122.121:24297
2012-08-20 13:17:03.303466 [INFO] mod_dingaling.c:1506 Stun Success 196.44.184.138:24297
2012-08-20 13:17:03.303466 [DEBUG] mod_dingaling.c:1520 Send video_rtcp Candidate 196.44.184.138:24297 [UWeSQTBFMScNn9ej]
2012-08-20 13:17:03.303466 [DEBUG] mod_dingaling.c:1560 Accepted 0 of 0 rtp candidates.
2012-08-20 13:17:03.303466 [DEBUG] mod_dingaling.c:1562 Accepted 0 of 0 rtcp candidates.
2012-08-20 13:17:03.303466 [DEBUG] mod_dingaling.c:1565 Accepted 0 of 0 video_rtp candidates
2012-08-20 13:17:03.303466 [DEBUG] mod_dingaling.c:1568 Accepted 0 of 0 video_rctp candidates
2012-08-20 13:17:03.743474 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:03.743474 [CRIT] libdingaling.c:1847 Sending packet 313 (2 left)

2012-08-20 13:17:03.743474 [NOTICE] libdingaling.c:1745 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="freedomfone01 at gmail.com/gtalkB846928D" to="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A" type="set" id="313">
  <ses:session xmlns:ses="http://www.google.com/session" type="candidates" id="c403880385" initiator="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gtalkB846928D">
    <ses:candidate name="rtp" address="196.44.184.138" port="24294" username="LHrfCI0Jp6gNld6x" password="hLOOtI5kAEk9dV8P" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="rtcp" address="196.44.184.138" port="24295" username="lhHB2Wam6U6KOPru" password="xOsvVbhEe6VIEWFP" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="video_rtp" address="196.44.184.138" port="24296" username="3bukkQnujVKIexCk" password="QdjcU5nIRjozAATu" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="video_rtcp" address="196.44.184.138" port="24297" username="UWeSQTBFMScNn9ej" password="P0HzgI3gGrKcgNF1" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
  </ses:session>
</iq>

2012-08-20 13:17:04.843477 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:05.943479 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:07.043454 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:08.163469 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:09.263454 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:09.263454 [CRIT] libdingaling.c:1847 Sending packet 313 (1 left)

2012-08-20 13:17:09.263454 [NOTICE] libdingaling.c:1745 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="freedomfone01 at gmail.com/gtalkB846928D" to="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A" type="set" id="313">
  <ses:session xmlns:ses="http://www.google.com/session" type="candidates" id="c403880385" initiator="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gtalkB846928D">
    <ses:candidate name="rtp" address="196.44.184.138" port="24294" username="LHrfCI0Jp6gNld6x" password="hLOOtI5kAEk9dV8P" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="rtcp" address="196.44.184.138" port="24295" username="lhHB2Wam6U6KOPru" password="xOsvVbhEe6VIEWFP" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="video_rtp" address="196.44.184.138" port="24296" username="3bukkQnujVKIexCk" password="QdjcU5nIRjozAATu" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="video_rtcp" address="196.44.184.138" port="24297" username="UWeSQTBFMScNn9ej" password="P0HzgI3gGrKcgNF1" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
  </ses:session>
</iq>

2012-08-20 13:17:10.363471 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:11.243449 [DEBUG] mod_dingaling.c:1560 Accepted 0 of 0 rtp candidates.
2012-08-20 13:17:11.243449 [DEBUG] mod_dingaling.c:1562 Accepted 0 of 0 rtcp candidates.
2012-08-20 13:17:11.243449 [DEBUG] mod_dingaling.c:1565 Accepted 0 of 0 video_rtp candidates
2012-08-20 13:17:11.243449 [DEBUG] mod_dingaling.c:1568 Accepted 0 of 0 video_rctp candidates
2012-08-20 13:17:11.463467 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:12.563463 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:13.663446 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:14.763484 [CRIT] libdingaling.c:1830 Processing 1 packets in retry queue

2012-08-20 13:17:14.763484 [CRIT] libdingaling.c:1847 Sending packet 313 (0 left)

2012-08-20 13:17:14.763484 [NOTICE] libdingaling.c:1745 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="freedomfone01 at gmail.com/gtalkB846928D" to="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gmail.3AF62E0A" type="set" id="313">
  <ses:session xmlns:ses="http://www.google.com/session" type="candidates" id="c403880385" initiator="2vys6x0b7wc1u2mu5wwcmalewq at public.talk.google.com/gtalkB846928D">
    <ses:candidate name="rtp" address="196.44.184.138" port="24294" username="LHrfCI0Jp6gNld6x" password="hLOOtI5kAEk9dV8P" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="rtcp" address="196.44.184.138" port="24295" username="lhHB2Wam6U6KOPru" password="xOsvVbhEe6VIEWFP" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="video_rtp" address="196.44.184.138" port="24296" username="3bukkQnujVKIexCk" password="QdjcU5nIRjozAATu" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="video_rtcp" address="196.44.184.138" port="24297" username="UWeSQTBFMScNn9ej" password="P0HzgI3gGrKcgNF1" preference="1,0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
  </ses:session>
</iq>

2012-08-20 13:17:14.763484 [CRIT] libdingaling.c:1856 Discarding packet 313

2012-08-20 13:17:21.243474 [DEBUG] mod_dingaling.c:1560 Accepted 0 of 0 rtp candidates.
2012-08-20 13:17:21.243474 [DEBUG] mod_dingaling.c:1562 Accepted 0 of 0 rtcp candidates.
2012-08-20 13:17:21.243474 [DEBUG] mod_dingaling.c:1565 Accepted 0 of 0 video_rtp candidates
2012-08-20 13:17:21.243474 [DEBUG] mod_dingaling.c:1568 Accepted 0 of 0 video_rctp candidates
2012-08-20 13:17:31.243462 [DEBUG] mod_dingaling.c:1560 Accepted 0 of 0 rtp candidates.
2012-08-20 13:17:31.243462 [DEBUG] mod_dingaling.c:1562 Accepted 0 of 0 rtcp candidates.
2012-08-20 13:17:31.243462 [DEBUG] mod_dingaling.c:1565 Accepted 0 of 0 video_rtp candidates
2012-08-20 13:17:31.243462 [DEBUG] mod_dingaling.c:1568 Accepted 0 of 0 video_rctp candidates
2012-08-20 13:17:41.243490 [DEBUG] mod_dingaling.c:1560 Accepted 0 of 0 rtp candidates.
2012-08-20 13:17:41.243490 [DEBUG] mod_dingaling.c:1562 Accepted 0 of 0 rtcp candidates.
2012-08-20 13:17:41.243490 [DEBUG] mod_dingaling.c:1565 Accepted 0 of 0 video_rtp candidates
2012-08-20 13:17:41.243490 [DEBUG] mod_dingaling.c:1568 Accepted 0 of 0 video_rctp candidates
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:1560 Accepted 0 of 0 rtp candidates.
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:1562 Accepted 0 of 0 rtcp candidates.
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:1565 Accepted 0 of 0 video_rtp candidates
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:1568 Accepted 0 of 0 video_rctp candidates
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:801 Terminate called from line 1783 state=CS_INIT
2012-08-20 13:17:51.243480 [DEBUG] switch_channel.c:2903 (DingaLing/new) Callstate Change DOWN -> HANGUP
2012-08-20 13:17:51.243480 [NOTICE] mod_dingaling.c:826 Hangup DingaLing/new [CS_INIT] [NORMAL_CLEARING]
2012-08-20 13:17:51.243480 [DEBUG] switch_channel.c:2926 Send signal DingaLing/new [KILL]
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:2019 DingaLing/new CHANNEL KILL
2012-08-20 13:17:51.243480 [DEBUG] switch_core_session.c:1229 Send signal DingaLing/new [BREAK]
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:2019 DingaLing/new CHANNEL KILL
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:424 (DingaLing/new) State INIT going to sleep
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:385 (DingaLing/new) Running State Change CS_HANGUP
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:625 (DingaLing/new) State HANGUP
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:1988 DingaLing/new CHANNEL HANGUP
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:47 DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:625 (DingaLing/new) State HANGUP going to sleep
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:416 (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING
2012-08-20 13:17:51.243480 [DEBUG] switch_core_session.c:1229 Send signal DingaLing/new [BREAK]
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:2019 DingaLing/new CHANNEL KILL
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:385 (DingaLing/new) Running State Change CS_REPORTING
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:685 (DingaLing/new) State REPORTING
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:79 DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:685 (DingaLing/new) State REPORTING going to sleep
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:410 (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY
2012-08-20 13:17:51.243480 [DEBUG] switch_core_session.c:1229 Send signal DingaLing/new [BREAK]
2012-08-20 13:17:51.243480 [DEBUG] mod_dingaling.c:2019 DingaLing/new CHANNEL KILL
2012-08-20 13:17:51.243480 [DEBUG] switch_core_session.c:1429 Session 7 (DingaLing/new) Locked, Waiting on external entities
2012-08-20 13:17:51.243480 [NOTICE] switch_core_session.c:1447 Session 7 (DingaLing/new) Ended
2012-08-20 13:17:51.243480 [NOTICE] switch_core_session.c:1449 Close Channel DingaLing/new [CS_DESTROY]
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:514 (DingaLing/new) Callstate Change HANGUP -> DOWN
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:517 (DingaLing/new) Running State Change CS_DESTROY
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:527 (DingaLing/new) State DESTROY
2012-08-20 13:17:51.243480 [CRIT] libdingaling.c:307 Destroyed Session c403880385

2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:86 DingaLing/new Standard DESTROY
2012-08-20 13:17:51.243480 [DEBUG] switch_core_state_machine.c:527 (DingaLing/new) State DESTROY going to sleep
freeswitch at internal> 
-------------- next part --------------
<include>
  <!-- Client Profile (Original mode) -->
  <!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
  <profile type="client">
   <param name="name" value="gmail.com"/>
   <param name="login" value="XXXXXXX at gmail.com/gtalk"/>
   <param name="password" value="XXXXXXXX"/>
   <param name="server" value="talk.google.com"/>
   <param name="dialplan" value="XML"/>
   <param name="context" value="default"/>
   <param name="message" value="FreedomFone IVR made easy!"/> <!-- You must specify a message for the profile to be valid -->
   <param name="rtp-ip" value="auto"/>
   <!-- if you are behind NAT, you should probably use STUN -->
   <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
   <param name="ext-rtp-ip" value="196.XXXXXXX"/>
   <param name="auto-login" value="true"/>
   <param name="auto-reply" value="Press *Call* to join my conference"/>
   <param name="sasl" value="plain"/>
   <param name="tls" value="true"/>
   <param name="use-rtp-timer" value="true"/>
   <param name="exten" value="5555"/> <!-- The extension you want a call in to connect to -->
   <param name="candidate-acl" value="wan.auto"/>
   <param name="local-network-acl" value="localnet.auto"/> 
  </profile>   
</include>

From vetali100 at gmail.com  Fri Sep  7 11:15:10 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Fri, 7 Sep 2012 00:15:10 -0700
Subject: [Freeswitch-users] More information on FreeSwitch
In-Reply-To: <CC20783F-329B-4377-914E-CE4D640529C6@gmail.com>
References: <CC20783F-329B-4377-914E-CE4D640529C6@gmail.com>
Message-ID: <CACc3D0XO5kOZ_imMy6XazzTGyYUpgJtrRRoAtkK9_WKTbFFo+w@mail.gmail.com>

You wrote that you do have Cookbook. But do you have the original
FreeSWITCH book? It has a lot of information.
Also, there is a mailing list which is quite alive.
Also, there is Cluecon (next will be in 2013)
What are your needs? And why editing files seems to be so scary?

Vitalie

2012/9/6 Stephan Swart <embedded65 at gmail.com>

> Hi
>
> Obviously i am new around here
> I am looking for more documentation on FreeSwitch
>
> I am aware of the Wiki
>
> I do have the cookbook
>
> What else is around
>
> I need to design a large voice deployment with Class 4 and Class 5
> switches.
>
> I need to know more detail and what is out there that can manage these
> devices in an easier manner than editing files. Administrators can be
> dangerous as well sometimes.
>
> Need some proper stability etc.
>
>
> Tanx
> Stephan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 avi at avimarcus.net  Fri Sep  7 11:28:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 7 Sep 2012 10:28:53 +0300
Subject: [Freeswitch-users] More information on FreeSwitch
In-Reply-To: <CACc3D0XO5kOZ_imMy6XazzTGyYUpgJtrRRoAtkK9_WKTbFFo+w@mail.gmail.com>
References: <CC20783F-329B-4377-914E-CE4D640529C6@gmail.com>
	<CACc3D0XO5kOZ_imMy6XazzTGyYUpgJtrRRoAtkK9_WKTbFFo+w@mail.gmail.com>
Message-ID: <CALNZuwnv1SzQ8FNC47U7SZuwAFo7u6JK3mxaGTmmOkVOfMF=rQ@mail.gmail.com>

>
> I need to know more detail and what is out there that can manage these
> devices in an easier manner than editing files.


See: http://wiki.freeswitch.org/wiki/Freeswitch_Gui

-Avi
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From evgeniy at bestnet.kharkov.ua  Fri Sep  7 11:59:57 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Fri, 07 Sep 2012 10:59:57 +0300
Subject: [Freeswitch-users] continue_on_fail problem
Message-ID: <5049A97D.9080907@bestnet.kharkov.ua>

Hi to all!
For example i have 2 phones: 7604503 and 7604504. If i'm calling to 
7604504 and it's busy, then call must be transfered to 7604503.
My dialplan is:

<context name="default">
<extension name="BESTNET">
<condition field="destination_number" expression="^(7604504)$">
<action application="set" data="dialed_extension=$1"/>
<action application="set" data="call_timeout=30"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=USER_BUSY"/>
<action application="set" data="tansfer_on_fail=USER_BUSY BUSY XML error"/>
<action application="bridge" 
data="user/${dialed_extension}@${domain_name}"/>
</condition>
</extension>
</context>
<context name="error">
<extension name="BUSY">
<condition field="${originate_disposition}" expression="^USER_BUSY$" 
continue="false" break="on-true">
<action application="log" data="IM HERE!!!!!!!!!!!!!!!!"/>
<action application="set" data="dialed_extension=$1"/>
<action application="set" data="call_timeout=30"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=false"/>
<action application="bridge" data="user/7604503@${domain_name}"/>
</condition>
</extension>
</context>

But it's doesn't work:( Please help and sorry for my english:)
-- 
Evgeniy Movlyan,
BestNet Ltd.


From sias at cpdata.co.za  Fri Sep  7 12:39:37 2012
From: sias at cpdata.co.za (Sias Mey)
Date: Fri, 7 Sep 2012 10:39:37 +0200 (SAST)
Subject: [Freeswitch-users] Confusion about sip hangup cause Q850 hangup
 cause and long struggles with a provider.
In-Reply-To: <1407731.17913.1347006256794.JavaMail.root@mail>
Message-ID: <22952124.18075.1347007177101.JavaMail.root@mail>

Hi, Im sure this subject has been beaten to death .. but many googles and many email searches hasent really managed to find me something.

Im a dev for a small company that writes call center software. Freeswitch was a godsend, thank you.

Now .. the confusion.

We are getting a lot of what seem to be strange hangup codes from a new provider big fights about loads of failled calls ensued blah blah.. much sip packet logging and manual inspection later.. I found the following.

from xmlcdr.


    <sip_hangup_disposition>recv_refuse</sip_hangup_disposition>                                                                                                                                                    
    <sip_term_status>408</sip_term_status>                                                                                                                                                                          
    <proto_specific_hangup_cause>sip%3A408</proto_specific_hangup_cause>                                                                                                                                            
    <sip_term_cause>102</sip_term_cause>                                                                                                                                                                            
    <hangup_cause>RECOVERY_ON_TIMER_EXPIRE</hangup_cause>                                                                                                                                                           
    <hangup_cause_q850>102</hangup_cause_q850>

this just to show its the same call
    <sip_call_id>0d4d4e76-735a-1230-d2ac-000423b5571b</sip_call_id>   

and from the sip messages.

SIP/2.0 408 Request Timeout
Call-ID: 0d4d4e76-735a-1230-d2ac-000423b5571b
Reason: Q.850;cause=18;text="no user responding"

And according to the very useful wiki page on Q.850 codes 408 should = 18 like it does in the providers response.

Why then is the q850 hangup cause in the CDR 102? and where does that translation come from.

This is a single example but I also have loads and loads where the CDR claims q850 code 18 but the sip messages provide 31 or a range of other codes.
I can understand if the q850 code from the sip message is not being read by FS since FS has to be a bit more agnostic than that and in the pas I have almost exclusively worked with direct connections to TDM hardware so my knowledge and understanding of the sip messages is rather limited. But even in that case, shouldent the q850 code in the cdr at least conform to the translation from the wiki page?

Oh I am not currently running the latest git release, having some libtiff issues on ubuntu to compile. I will respond to this again if I manage that and it helps matters.

Thank you for your time and help,
Regards
Sias
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From fdelawarde at wirelessmundi.com  Fri Sep  7 12:42:36 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 07 Sep 2012 10:42:36 +0200
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <5049423B.9070401@communicatefreely.net>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
Message-ID: <1347007356.19904.43.camel@luna.madrid.commsmundi.com>

From my experience with a some Aastra models (mainly 6751i, 6755i and
6757i), users complain about usability issues, even for simple tasks
like call transfer, and admins complain for too many bugs.

Overall sound quality is good, bugs are little annoying things, for
example you can't change the time-zone from the Web-UI and you need to
physically go to the phone menu; or configs that get screwed after any
firmware update (need to reset to factory and re-provision).

We also had bad luck with the 51i, which became EOL before many bad bugs
got the chance to be corrected. I had to change 5 out of 20 so far, as
they slowly started dying after a couple of years (blank screen, or
unable to provision at some point). The 55i and 57i still have firmware
updates and are getting a bit better each time.

I find Yealink phones have a better overall quality (robustness,
interop, sound, and USABILITY) for almost half the price, heck I even
find latest Grandstream firmwares to be better than Aastra's.

Fran?ois.


On Thu, 2012-09-06 at 20:39 -0400, Tim St. Pierre wrote: 
> Yes, those were terrible.  We sold four of them, and have since replaced 
> them all.
> 
> I think they were part of some acquisition that Aastra made, and they 
> never really matured.  I wouldn't judge Aastra by that phone.
> 
> -Tim
> 
> Emrah wrote:
> > That's great Tim, thanks! I'll check them out.
> > I purchased 3 Aastra dect phones for my apt (some clone of Snom M3) and was super disappointed by the quality? The speakerphone is simply unusable.
> >
> > Regards,
> > Emrah
> > On Sep 6, 2012, at 7:45 PM, "Tim St. Pierre" <fs-list at communicatefreely.net> wrote:
> >
> >   
> >> Hi Emrah,
> >>
> >> I have been using Aastra phones extensively.  If you stay with the 5xi 
> >> series, you should get good sound quality.  The 6739i is really nice, 
> >> but touch screen, so that's out.
> >>
> >> There are a few different ways to provision them - the simplest way is 
> >> for them to download a config file (ours are rendered on the fly with 
> >> PHP).  You can also use the Web UI, or send the phone XML commands.  If 
> >> you make changes in the web UI or push config via XML, the changes take 
> >> place right away on the newer firmwares.  This is great for 
> >> experimenting, as you can change SIP settings around without having to 
> >> reboot every time.  You can also interface the phones to other systems 
> >> using the XML interface, or via some proprietary software.  There is a 
> >> third party app that lets you click to dial from your computer, and you 
> >> get on-screen notifications of incoming calls.
> >>
> >> Some versions of firmware are buggy, but they are getting better.  I'm 
> >> waiting for IPv6 compatible firmware, but I'm told that it's in the 
> >> works and they put me on the beta test list.
> >>
> >> Hope that's helpful.
> >>
> >> -Tim
> >>
> >> Emrah wrote:
> >>     
> >>> Hi all,
> >>>
> >>> I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.
> >>>
> >>> 1. It's touch screen and I'm blind.
> >>> 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
> >>> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.
> >>>
> >>> I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.
> >>>
> >>> What can you recommend that is stable, versatile, open and good quality? 
> >>>
> >>> Cheers,
> >>> 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
> >>>
> >>>       
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>     
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >   
> Y
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep  7 13:59:23 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 7 Sep 2012 05:59:23 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <1347007356.19904.43.camel@luna.madrid.commsmundi.com>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
Message-ID: <F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>

Thanks for that!
You are giving me the impression that Polycom is years ahead. :)

Best,
Emrah
On Sep 7, 2012, at 4:42 AM, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:

> From my experience with a some Aastra models (mainly 6751i, 6755i and
> 6757i), users complain about usability issues, even for simple tasks
> like call transfer, and admins complain for too many bugs.
> 
> Overall sound quality is good, bugs are little annoying things, for
> example you can't change the time-zone from the Web-UI and you need to
> physically go to the phone menu; or configs that get screwed after any
> firmware update (need to reset to factory and re-provision).
> 
> We also had bad luck with the 51i, which became EOL before many bad bugs
> got the chance to be corrected. I had to change 5 out of 20 so far, as
> they slowly started dying after a couple of years (blank screen, or
> unable to provision at some point). The 55i and 57i still have firmware
> updates and are getting a bit better each time.
> 
> I find Yealink phones have a better overall quality (robustness,
> interop, sound, and USABILITY) for almost half the price, heck I even
> find latest Grandstream firmwares to be better than Aastra's.
> 
> Fran?ois.
> 
> 
> On Thu, 2012-09-06 at 20:39 -0400, Tim St. Pierre wrote: 
>> Yes, those were terrible.  We sold four of them, and have since replaced 
>> them all.
>> 
>> I think they were part of some acquisition that Aastra made, and they 
>> never really matured.  I wouldn't judge Aastra by that phone.
>> 
>> -Tim
>> 
>> Emrah wrote:
>>> That's great Tim, thanks! I'll check them out.
>>> I purchased 3 Aastra dect phones for my apt (some clone of Snom M3) and was super disappointed by the quality? The speakerphone is simply unusable.
>>> 
>>> Regards,
>>> Emrah
>>> On Sep 6, 2012, at 7:45 PM, "Tim St. Pierre" <fs-list at communicatefreely.net> wrote:
>>> 
>>> 
>>>> Hi Emrah,
>>>> 
>>>> I have been using Aastra phones extensively.  If you stay with the 5xi 
>>>> series, you should get good sound quality.  The 6739i is really nice, 
>>>> but touch screen, so that's out.
>>>> 
>>>> There are a few different ways to provision them - the simplest way is 
>>>> for them to download a config file (ours are rendered on the fly with 
>>>> PHP).  You can also use the Web UI, or send the phone XML commands.  If 
>>>> you make changes in the web UI or push config via XML, the changes take 
>>>> place right away on the newer firmwares.  This is great for 
>>>> experimenting, as you can change SIP settings around without having to 
>>>> reboot every time.  You can also interface the phones to other systems 
>>>> using the XML interface, or via some proprietary software.  There is a 
>>>> third party app that lets you click to dial from your computer, and you 
>>>> get on-screen notifications of incoming calls.
>>>> 
>>>> Some versions of firmware are buggy, but they are getting better.  I'm 
>>>> waiting for IPv6 compatible firmware, but I'm told that it's in the 
>>>> works and they put me on the beta test list.
>>>> 
>>>> Hope that's helpful.
>>>> 
>>>> -Tim
>>>> 
>>>> Emrah wrote:
>>>> 
>>>>> Hi all,
>>>>> 
>>>>> I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.
>>>>> 
>>>>> 1. It's touch screen and I'm blind.
>>>>> 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
>>>>> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.
>>>>> 
>>>>> I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.
>>>>> 
>>>>> What can you recommend that is stable, versatile, open and good quality? 
>>>>> 
>>>>> Cheers,
>>>>> 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
>>>>> 
>>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> Y
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Sep  7 14:21:21 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Fri, 7 Sep 2012 10:21:21 +0000
Subject: [Freeswitch-users] mod_shout / telecast
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2950A2@exchange.besharp.at>

Hi,

I'm trying to embed the mp3 link from http://x.x.x.x:8080/webapi/telecast/mp3/72d6e296-f8d4-11e1-b7b6-f1154fdbf73a/2001.mp3
(from http://x.x.x.x:8080/webapi/telecast/index ) into an embedded player object in a html page.
(i'm on latest git)

The problem: i'm getting garbage as output from the above link, regardless which browser i use.
Even winamp doesnt recognize the stream.

Is mod_shout / telecast currently broken or am i doing something wrong ?

Any help appreciated,

thanks
gw
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From qasimakhan at gmail.com  Fri Sep  7 14:52:20 2012
From: qasimakhan at gmail.com (qasimakhan at gmail.com)
Date: Fri, 7 Sep 2012 15:52:20 +0500
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
Message-ID: <CAPBKugv26biiRTqSgAB1XsmsSRJ=eKAFQ73OOt2MRAL50tdpRw@mail.gmail.com>

I have been using Both Polycom and Cisco at my work. Both are good, There
are firmwares available on some cisco models on which you can configure
your phone from firmware as well as auto provisioning. There were however a
few bugs in polycom here and there, Not that much annoying though. Over all
considering quality and usability i would recommend cisco.

Regards,
Qasim

On Fri, Sep 7, 2012 at 2:59 PM, Emrah <lists at kavun.ch> wrote:

> Thanks for that!
> You are giving me the impression that Polycom is years ahead. :)
>
> Best,
> Emrah
> On Sep 7, 2012, at 4:42 AM, Fran?ois Delawarde <
> fdelawarde at wirelessmundi.com> wrote:
>
> > From my experience with a some Aastra models (mainly 6751i, 6755i and
> > 6757i), users complain about usability issues, even for simple tasks
> > like call transfer, and admins complain for too many bugs.
> >
> > Overall sound quality is good, bugs are little annoying things, for
> > example you can't change the time-zone from the Web-UI and you need to
> > physically go to the phone menu; or configs that get screwed after any
> > firmware update (need to reset to factory and re-provision).
> >
> > We also had bad luck with the 51i, which became EOL before many bad bugs
> > got the chance to be corrected. I had to change 5 out of 20 so far, as
> > they slowly started dying after a couple of years (blank screen, or
> > unable to provision at some point). The 55i and 57i still have firmware
> > updates and are getting a bit better each time.
> >
> > I find Yealink phones have a better overall quality (robustness,
> > interop, sound, and USABILITY) for almost half the price, heck I even
> > find latest Grandstream firmwares to be better than Aastra's.
> >
> > Fran?ois.
> >
> >
> > On Thu, 2012-09-06 at 20:39 -0400, Tim St. Pierre wrote:
> >> Yes, those were terrible.  We sold four of them, and have since replaced
> >> them all.
> >>
> >> I think they were part of some acquisition that Aastra made, and they
> >> never really matured.  I wouldn't judge Aastra by that phone.
> >>
> >> -Tim
> >>
> >> Emrah wrote:
> >>> That's great Tim, thanks! I'll check them out.
> >>> I purchased 3 Aastra dect phones for my apt (some clone of Snom M3)
> and was super disappointed by the quality? The speakerphone is simply
> unusable.
> >>>
> >>> Regards,
> >>> Emrah
> >>> On Sep 6, 2012, at 7:45 PM, "Tim St. Pierre" <
> fs-list at communicatefreely.net> wrote:
> >>>
> >>>
> >>>> Hi Emrah,
> >>>>
> >>>> I have been using Aastra phones extensively.  If you stay with the 5xi
> >>>> series, you should get good sound quality.  The 6739i is really nice,
> >>>> but touch screen, so that's out.
> >>>>
> >>>> There are a few different ways to provision them - the simplest way is
> >>>> for them to download a config file (ours are rendered on the fly with
> >>>> PHP).  You can also use the Web UI, or send the phone XML commands.
>  If
> >>>> you make changes in the web UI or push config via XML, the changes
> take
> >>>> place right away on the newer firmwares.  This is great for
> >>>> experimenting, as you can change SIP settings around without having to
> >>>> reboot every time.  You can also interface the phones to other systems
> >>>> using the XML interface, or via some proprietary software.  There is a
> >>>> third party app that lets you click to dial from your computer, and
> you
> >>>> get on-screen notifications of incoming calls.
> >>>>
> >>>> Some versions of firmware are buggy, but they are getting better.  I'm
> >>>> waiting for IPv6 compatible firmware, but I'm told that it's in the
> >>>> works and they put me on the beta test list.
> >>>>
> >>>> Hope that's helpful.
> >>>>
> >>>> -Tim
> >>>>
> >>>> Emrah wrote:
> >>>>
> >>>>> Hi all,
> >>>>>
> >>>>> I have tried many IP phones and nothing comes close to the audio
> quality of a Polycom. I use a VVX1500 and everything is top notch, but it
> doesn't suit me entirely.
> >>>>>
> >>>>> 1. It's touch screen and I'm blind.
> >>>>> 2. It's always a pain to restart the phone anytime you make a change
> in the provisioning configs. It's great in a corporate environment, but not
> so much for experimenting.
> >>>>> 3. There isn't many ways to interface with the phone? E.g.: trigger
> a dial out from your computer's address book.
> >>>>>
> >>>>> I tried Snom a couple years ago and the audio quality was not good,
> with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty
> much out of the question because of their closed configs and Linksys is
> pretty low end if my recollection is correct.
> >>>>>
> >>>>> What can you recommend that is stable, versatile, open and good
> quality?
> >>>>>
> >>>>> Cheers,
> >>>>> 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
> >>>>>
> >>>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >> Y
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fdelawarde at wirelessmundi.com  Fri Sep  7 15:07:25 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 07 Sep 2012 13:07:25 +0200
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
Message-ID: <1347016045.19904.64.camel@luna.madrid.commsmundi.com>

I have to say I'm very impressed with the Yealink T28P, comparable to
Polycom 650 in terms of quality and features, at 1/2 price. It's worth a
try.


On Fri, 2012-09-07 at 05:59 -0400, Emrah wrote: 
> Thanks for that!
> You are giving me the impression that Polycom is years ahead. :)
> 
> Best,
> Emrah
> On Sep 7, 2012, at 4:42 AM, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:
> 
> > From my experience with a some Aastra models (mainly 6751i, 6755i and
> > 6757i), users complain about usability issues, even for simple tasks
> > like call transfer, and admins complain for too many bugs.
> > 
> > Overall sound quality is good, bugs are little annoying things, for
> > example you can't change the time-zone from the Web-UI and you need to
> > physically go to the phone menu; or configs that get screwed after any
> > firmware update (need to reset to factory and re-provision).
> > 
> > We also had bad luck with the 51i, which became EOL before many bad bugs
> > got the chance to be corrected. I had to change 5 out of 20 so far, as
> > they slowly started dying after a couple of years (blank screen, or
> > unable to provision at some point). The 55i and 57i still have firmware
> > updates and are getting a bit better each time.
> > 
> > I find Yealink phones have a better overall quality (robustness,
> > interop, sound, and USABILITY) for almost half the price, heck I even
> > find latest Grandstream firmwares to be better than Aastra's.
> > 
> > Fran?ois.
> > 
> > 
> > On Thu, 2012-09-06 at 20:39 -0400, Tim St. Pierre wrote: 
> >> Yes, those were terrible.  We sold four of them, and have since replaced 
> >> them all.
> >> 
> >> I think they were part of some acquisition that Aastra made, and they 
> >> never really matured.  I wouldn't judge Aastra by that phone.
> >> 
> >> -Tim
> >> 
> >> Emrah wrote:
> >>> That's great Tim, thanks! I'll check them out.
> >>> I purchased 3 Aastra dect phones for my apt (some clone of Snom M3) and was super disappointed by the quality? The speakerphone is simply unusable.
> >>> 
> >>> Regards,
> >>> Emrah
> >>> On Sep 6, 2012, at 7:45 PM, "Tim St. Pierre" <fs-list at communicatefreely.net> wrote:
> >>> 
> >>> 
> >>>> Hi Emrah,
> >>>> 
> >>>> I have been using Aastra phones extensively.  If you stay with the 5xi 
> >>>> series, you should get good sound quality.  The 6739i is really nice, 
> >>>> but touch screen, so that's out.
> >>>> 
> >>>> There are a few different ways to provision them - the simplest way is 
> >>>> for them to download a config file (ours are rendered on the fly with 
> >>>> PHP).  You can also use the Web UI, or send the phone XML commands.  If 
> >>>> you make changes in the web UI or push config via XML, the changes take 
> >>>> place right away on the newer firmwares.  This is great for 
> >>>> experimenting, as you can change SIP settings around without having to 
> >>>> reboot every time.  You can also interface the phones to other systems 
> >>>> using the XML interface, or via some proprietary software.  There is a 
> >>>> third party app that lets you click to dial from your computer, and you 
> >>>> get on-screen notifications of incoming calls.
> >>>> 
> >>>> Some versions of firmware are buggy, but they are getting better.  I'm 
> >>>> waiting for IPv6 compatible firmware, but I'm told that it's in the 
> >>>> works and they put me on the beta test list.
> >>>> 
> >>>> Hope that's helpful.
> >>>> 
> >>>> -Tim
> >>>> 
> >>>> Emrah wrote:
> >>>> 
> >>>>> Hi all,
> >>>>> 
> >>>>> I have tried many IP phones and nothing comes close to the audio quality of a Polycom. I use a VVX1500 and everything is top notch, but it doesn't suit me entirely.
> >>>>> 
> >>>>> 1. It's touch screen and I'm blind.
> >>>>> 2. It's always a pain to restart the phone anytime you make a change in the provisioning configs. It's great in a corporate environment, but not so much for experimenting.
> >>>>> 3. There isn't many ways to interface with the phone? E.g.: trigger a dial out from your computer's address book.
> >>>>> 
> >>>>> I tried Snom a couple years ago and the audio quality was not good, with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty much out of the question because of their closed configs and Linksys is pretty low end if my recollection is correct.
> >>>>> 
> >>>>> What can you recommend that is stable, versatile, open and good quality? 
> >>>>> 
> >>>>> Cheers,
> >>>>> 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
> >>>>> 
> >>>>> 
> >>>> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>> 
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>> 
> >>> 
> >>> 
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>> 
> >>> 
> >>> 
> >>> 
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>> 
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>> 
> >> Y
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> 
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 thewinelake.com  Fri Sep  7 15:27:46 2012
From: alex at thewinelake.com (Alex Lake)
Date: Fri, 7 Sep 2012 12:27:46 +0100
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <5049432F.3090406@communicatefreely.net>
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>
	<5049432F.3090406@communicatefreely.net>
Message-ID: <2C76AFC8-EB11-4218-A7F4-FE8B2BB6CC43@digitalmail.com>

No... A vicious circle, I suspect!



On 7 Sep 2012, at 01:43, "Tim St. Pierre" <fs-list at communicatefreely.net> wrote:

> Hi Alex,
> 
> I see that this is an older thread, but I now have native IPv6 
> connectivity set up at our datacenter, tunneled IPv6 at many customer 
> sites with native on the way.
> 
> I agree that it gets rid of all the NAT issues, so I'm hoping to try it 
> out very soon.  For larger installations, I'm insisting that native IPv6 
> be available within the next few months.
> 
> I'm having a hard time finding phones that can do IPv6 though.
> 
> Any suggestions?
> 
> 
> 
> Alex 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
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Sep  7 15:40:20 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Fri, 7 Sep 2012 11:40:20 +0000
Subject: [Freeswitch-users] mod_shout / telecast
In-Reply-To: <D1925DECD316894DAADB84076C1B7D827E2950A2@exchange.besharp.at>
References: <D1925DECD316894DAADB84076C1B7D827E2950A2@exchange.besharp.at>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E295143@exchange.besharp.at>

I answer that myself:
There is a jira for this, http://jira.freeswitch.org/browse/FS-3960
Seems like /webapi produces some other output than /api

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gerald Weber
Gesendet: Freitag, 07. September 2012 12:21
An: freeswitch-users at lists.freeswitch.org
Betreff: [Freeswitch-users] mod_shout / telecast

Hi,

I'm trying to embed the mp3 link from http://x.x.x.x:8080/webapi/telecast/mp3/72d6e296-f8d4-11e1-b7b6-f1154fdbf73a/2001.mp3
(from http://x.x.x.x:8080/webapi/telecast/index ) into an embedded player object in a html page.
(i'm on latest git)

The problem: i'm getting garbage as output from the above link, regardless which browser i use.
Even winamp doesnt recognize the stream.

Is mod_shout / telecast currently broken or am i doing something wrong ?

Any help appreciated,

thanks
gw

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From dujinfang at gmail.com  Fri Sep  7 15:47:20 2012
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 7 Sep 2012 19:47:20 +0800
Subject: [Freeswitch-users] mod_shout / telecast
In-Reply-To: <D1925DECD316894DAADB84076C1B7D827E295143@exchange.besharp.at>
References: <D1925DECD316894DAADB84076C1B7D827E2950A2@exchange.besharp.at>
	<D1925DECD316894DAADB84076C1B7D827E295143@exchange.besharp.at>
Message-ID: <A27FE7A21D714632A76C01A041E66A51@gmail.com>

- apply the patch and see if it helps  

- comment on the jira to get more attention  

--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Friday, September 7, 2012 at 7:40 PM, Gerald Weber wrote:

> I answer that myself:
> There is a jira for this, http://jira.freeswitch.org/browse/FS-3960
> Seems like /webapi produces some other output than /api
>   
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gerald Weber
> Gesendet: Freitag, 07. September 2012 12:21
> An: freeswitch-users at lists.freeswitch.org (mailto:freeswitch-users at lists.freeswitch.org)
> Betreff: [Freeswitch-users] mod_shout / telecast
>   
> Hi,
>   
> I?m trying to embed the mp3 link from http://x.x.x.x:8080/webapi/telecast/mp3/72d6e296-f8d4-11e1-b7b6-f1154fdbf73a/2001.mp3  
> (from http://x.x.x.x:8080/webapi/telecast/index ) into an embedded player object in a html page.
> (i?m on latest git)
>   
> The problem: i?m getting garbage as output from the above link, regardless which browser i use.
> Even winamp doesnt recognize the stream.
>   
> Is mod_shout / telecast currently broken or am i doing something wrong ?
>   
> Any help appreciated,
>   
> thanks
> gw
>   
>  
>  
> _________________________________________________________________________
> 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 freeswitch-list at puzzled.xs4all.nl  Fri Sep  7 15:54:47 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 07 Sep 2012 13:54:47 +0200
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <CAPBKugv26biiRTqSgAB1XsmsSRJ=eKAFQ73OOt2MRAL50tdpRw@mail.gmail.com>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
	<CAPBKugv26biiRTqSgAB1XsmsSRJ=eKAFQ73OOt2MRAL50tdpRw@mail.gmail.com>
Message-ID: <5049E087.3060207@puzzled.xs4all.nl>

On 09/07/2012 12:52 PM, qasimakhan at gmail.com wrote:
[snip]
> Over all considering quality and usability i would
> recommend cisco.

SIP seems to be a second rate citizen at Cisco where SCCP is king. If 
you deploy Cisco's own switch (CallManager/CUCM or whatever it's called 
these days) with their phones with SCCP firmware you should be good to 
go. However, I would not use Cisco IP phones with a non-Cisco SIP switch 
when there are alternatives available like Polycom, Yealink, Aastra, 
Snom, Grandstream (in order of preference).

Regards,
Patrick



From sdevoy at bizfocused.com  Fri Sep  7 15:58:47 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 7 Sep 2012 07:58:47 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <CAPBKugv26biiRTqSgAB1XsmsSRJ=eKAFQ73OOt2MRAL50tdpRw@mail.gmail.com>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>	<504935A0.9070106@communicatefreely.net>	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>	<5049423B.9070401@communicatefreely.net>	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
	<CAPBKugv26biiRTqSgAB1XsmsSRJ=eKAFQ73OOt2MRAL50tdpRw@mail.gmail.com>
Message-ID: <019501cd8cf0$2372d3e0$6a587ba0$@bizfocused.com>

We have also decided to be a CISCO 3xx/5xx  series shop (and Cisco/Linksys
ATAs).  They just work great.  They are not the cheapest, but we think the
quality justifies the cost.

 

Regards,

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
qasimakhan at gmail.com
Sent: Friday, September 07, 2012 6:52 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Best SIP phone?

 

I have been using Both Polycom and Cisco at my work. Both are good, There
are firmwares available on some cisco models on which you can configure your
phone from firmware as well as auto provisioning. There were however a few
bugs in polycom here and there, Not that much annoying though. Over all
considering quality and usability i would recommend cisco. 

Regards,
Qasim

On Fri, Sep 7, 2012 at 2:59 PM, Emrah <lists at kavun.ch> wrote:

Thanks for that!
You are giving me the impression that Polycom is years ahead. :)

Best,
Emrah

On Sep 7, 2012, at 4:42 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:

> From my experience with a some Aastra models (mainly 6751i, 6755i and
> 6757i), users complain about usability issues, even for simple tasks
> like call transfer, and admins complain for too many bugs.
>
> Overall sound quality is good, bugs are little annoying things, for
> example you can't change the time-zone from the Web-UI and you need to
> physically go to the phone menu; or configs that get screwed after any
> firmware update (need to reset to factory and re-provision).
>
> We also had bad luck with the 51i, which became EOL before many bad bugs
> got the chance to be corrected. I had to change 5 out of 20 so far, as
> they slowly started dying after a couple of years (blank screen, or
> unable to provision at some point). The 55i and 57i still have firmware
> updates and are getting a bit better each time.
>
> I find Yealink phones have a better overall quality (robustness,
> interop, sound, and USABILITY) for almost half the price, heck I even
> find latest Grandstream firmwares to be better than Aastra's.
>
> Fran?ois.
>
>
> On Thu, 2012-09-06 at 20:39 -0400, Tim St. Pierre wrote:
>> Yes, those were terrible.  We sold four of them, and have since replaced
>> them all.
>>
>> I think they were part of some acquisition that Aastra made, and they
>> never really matured.  I wouldn't judge Aastra by that phone.
>>
>> -Tim
>>
>> Emrah wrote:
>>> That's great Tim, thanks! I'll check them out.
>>> I purchased 3 Aastra dect phones for my apt (some clone of Snom M3) and
was super disappointed by the quality
 The speakerphone is simply unusable.
>>>
>>> Regards,
>>> Emrah
>>> On Sep 6, 2012, at 7:45 PM, "Tim St. Pierre"
<fs-list at communicatefreely.net> wrote:
>>>
>>>
>>>> Hi Emrah,
>>>>
>>>> I have been using Aastra phones extensively.  If you stay with the 5xi
>>>> series, you should get good sound quality.  The 6739i is really nice,
>>>> but touch screen, so that's out.
>>>>
>>>> There are a few different ways to provision them - the simplest way is
>>>> for them to download a config file (ours are rendered on the fly with
>>>> PHP).  You can also use the Web UI, or send the phone XML commands.  If
>>>> you make changes in the web UI or push config via XML, the changes take
>>>> place right away on the newer firmwares.  This is great for
>>>> experimenting, as you can change SIP settings around without having to
>>>> reboot every time.  You can also interface the phones to other systems
>>>> using the XML interface, or via some proprietary software.  There is a
>>>> third party app that lets you click to dial from your computer, and you
>>>> get on-screen notifications of incoming calls.
>>>>
>>>> Some versions of firmware are buggy, but they are getting better.  I'm
>>>> waiting for IPv6 compatible firmware, but I'm told that it's in the
>>>> works and they put me on the beta test list.
>>>>
>>>> Hope that's helpful.
>>>>
>>>> -Tim
>>>>
>>>> Emrah wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> I have tried many IP phones and nothing comes close to the audio
quality of a Polycom. I use a VVX1500 and everything is top notch, but it
doesn't suit me entirely.
>>>>>
>>>>> 1. It's touch screen and I'm blind.
>>>>> 2. It's always a pain to restart the phone anytime you make a change
in the provisioning configs. It's great in a corporate environment, but not
so much for experimenting.
>>>>> 3. There isn't many ways to interface with the phone
 E.g.: trigger a
dial out from your computer's address book.
>>>>>
>>>>> I tried Snom a couple years ago and the audio quality was not good,
with plenty of echo on speakerphone and buggy firmwares. Cisco is pretty
much out of the question because of their closed configs and Linksys is
pretty low end if my recollection is correct.
>>>>>
>>>>> What can you recommend that is stable, versatile, open and good
quality?
>>>>>
>>>>> Cheers,
>>>>> 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
>>>>>
>>>>>
>>>>
_________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
_________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> Y
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Fri Sep  7 16:01:08 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 7 Sep 2012 08:01:08 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <5049E087.3060207@puzzled.xs4all.nl>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>	<504935A0.9070106@communicatefreely.net>	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>	<5049423B.9070401@communicatefreely.net>	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>	<CAPBKugv26biiRTqSgAB1XsmsSRJ=eKAFQ73OOt2MRAL50tdpRw@mail.gmail.com>
	<5049E087.3060207@puzzled.xs4all.nl>
Message-ID: <019a01cd8cf0$778c1040$66a430c0$@bizfocused.com>

I agree with regard to the Cisco 7000 series, but the 300/500 series phones
are totally SIP friendly.

Regards,
Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Patrick
Lists
Sent: Friday, September 07, 2012 7:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Best SIP phone?

On 09/07/2012 12:52 PM, qasimakhan at gmail.com wrote:
[snip]
> Over all considering quality and usability i would recommend cisco.

SIP seems to be a second rate citizen at Cisco where SCCP is king. If you
deploy Cisco's own switch (CallManager/CUCM or whatever it's called these
days) with their phones with SCCP firmware you should be good to go.
However, I would not use Cisco IP phones with a non-Cisco SIP switch when
there are alternatives available like Polycom, Yealink, Aastra, Snom,
Grandstream (in order of preference).

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 freeswitch-list at puzzled.xs4all.nl  Fri Sep  7 16:12:34 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 07 Sep 2012 14:12:34 +0200
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <1347016045.19904.64.camel@luna.madrid.commsmundi.com>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
	<1347016045.19904.64.camel@luna.madrid.commsmundi.com>
Message-ID: <5049E4B2.5090205@puzzled.xs4all.nl>

Hi Fran?ois,

On 09/07/2012 01:07 PM, Fran?ois Delawarde wrote:
> I have to say I'm very impressed with the Yealink T28P, comparable to
> Polycom 650 in terms of quality and features, at 1/2 price. It's worth a
> try.

Quick question I could not find on the Yealink website or in the 
datasheet: is the screen of the T28P backlit? And the keys (like 0-9)?

Thanks,
Patrick



From gerald.weber at besharp.at  Fri Sep  7 16:29:22 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Fri, 7 Sep 2012 12:29:22 +0000
Subject: [Freeswitch-users] mod_shout / telecast
In-Reply-To: <A27FE7A21D714632A76C01A041E66A51@gmail.com>
References: <D1925DECD316894DAADB84076C1B7D827E2950A2@exchange.besharp.at>
	<D1925DECD316894DAADB84076C1B7D827E295143@exchange.besharp.at>
	<A27FE7A21D714632A76C01A041E66A51@gmail.com>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2951B3@exchange.besharp.at>

I applied the patch before i wrote my last mail, and it works fine.
I commented and voted on jira now ?


Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Seven Du
Gesendet: Freitag, 07. September 2012 13:47
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_shout / telecast

- apply the patch and see if it helps

- comment on the jira to get more attention

--
Seven Du
Sent with Sparrow<http://www.sparrowmailapp.com/?sig>


On Friday, September 7, 2012 at 7:40 PM, Gerald Weber wrote:

I answer that myself:

There is a jira for this, http://jira.freeswitch.org/browse/FS-3960

Seems like /webapi produces some other output than /api



Von: 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 Gerald Weber
Gesendet: Freitag, 07. September 2012 12:21
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Betreff: [Freeswitch-users] mod_shout / telecast



Hi,



I?m trying to embed the mp3 link from http://x.x.x.x:8080/webapi/telecast/mp3/72d6e296-f8d4-11e1-b7b6-f1154fdbf73a/2001.mp3

(from http://x.x.x.x:8080/webapi/telecast/index ) into an embedded player object in a html page.

(i?m on latest git)



The problem: i?m getting garbage as output from the above link, regardless which browser i use.

Even winamp doesnt recognize the stream.



Is mod_shout / telecast currently broken or am i doing something wrong ?



Any help appreciated,



thanks

gw


_________________________________________________________________________
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
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From fdelawarde at wirelessmundi.com  Fri Sep  7 17:24:59 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 07 Sep 2012 15:24:59 +0200
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <5049E4B2.5090205@puzzled.xs4all.nl>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
	<1347016045.19904.64.camel@luna.madrid.commsmundi.com>
	<5049E4B2.5090205@puzzled.xs4all.nl>
Message-ID: <1347024299.19904.72.camel@luna.madrid.commsmundi.com>

Hi Patrick,

Not sure I understand the question. Aren't all LCD screens backlit?

But yes the screen is a 320x160 LCD with backlight. About keys, you have
6 programmable keys on the top part and 10 more on the right with LEDs
for BLF or other things. The other keys (dtmf, navigation, volume, etc.)
have no light.

Fran?ois.


On Fri, 2012-09-07 at 14:12 +0200, Patrick Lists wrote: 
> Hi Fran?ois,
> 
> On 09/07/2012 01:07 PM, Fran?ois Delawarde wrote:
> > I have to say I'm very impressed with the Yealink T28P, comparable to
> > Polycom 650 in terms of quality and features, at 1/2 price. It's worth a
> > try.
> 
> Quick question I could not find on the Yealink website or in the 
> datasheet: is the screen of the T28P backlit? And the keys (like 0-9)?
> 
> Thanks,
> 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 bclark at grasshopper.com  Fri Sep  7 17:43:42 2012
From: bclark at grasshopper.com (Brett Clark - Grasshopper)
Date: Fri, 7 Sep 2012 08:43:42 -0500
Subject: [Freeswitch-users] best FS interface for development
Message-ID: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>

There seems to be at least 4 general ways for customizing FS:
- Use the event socket and build an app to interface with it
- Use mod lua, or similar, to develop in a particular language
- Develop your own 'module' to interface directly with the core
- Use the XML interface to implement IVR, PBX, and voicemail type functionality and all the rest.

How prevalent is each approach in the community?  It seems like everyone is doing a little of both, but maybe someone has a better handle of what the actual ratios are?

Which is best suited for an arbitrarily large and complex application?  I realize this isn't a simple question-what I want is to understand the most evolved and featureful way to interface with FS?  I don't want to adopt an approach which isn't being actively maintained or is missing features.  As new stuff is added to FS, which approach will allow me to adopt those new features most easily?

Thanks!
Brett





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From ben at langfeld.co.uk  Fri Sep  7 18:00:50 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 7 Sep 2012 16:00:50 +0200
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
Message-ID: <CAAyX+KFL-vjBODeqVG190wxenNu=gPjDoZOvJ=TOmu8YfzS84w@mail.gmail.com>

The Adhearsion framework firmly takes the 3PCC approach via inbound event
socket, and does a similar thing on Asterisk. We believe that to be the
best way to isolate business logic from the switch.

Regards,
Ben Langfeld


On 7 September 2012 15:43, Brett Clark - Grasshopper <bclark at grasshopper.com
> wrote:

> There seems to be at least 4 general ways for customizing FS:****
>
> - Use the event socket and build an app to interface with it****
>
> - Use mod lua, or similar, to develop in a particular language****
>
> - Develop your own ?module? to interface directly with the core****
>
> - Use the XML interface to implement IVR, PBX, and voicemail type
> functionality and all the rest.****
>
> ** **
>
> How prevalent is each approach in the community?  It seems like everyone
> is doing a little of both, but maybe someone has a better handle of what
> the actual ratios are?  ****
>
> ** **
>
> Which is best suited for an arbitrarily large and complex application?  I
> realize this isn?t a simple question?what I want is to understand the most
> evolved and featureful way to interface with FS?  I don?t want to adopt an
> approach which isn?t being actively maintained or is missing features.  As
> new stuff is added to FS, which approach will allow me to adopt those new
> features most easily?****
>
> ** **
>
> Thanks!
> Brett****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep  7 18:39:40 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 07 Sep 2012 16:39:40 +0200
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <1347024299.19904.72.camel@luna.madrid.commsmundi.com>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
	<1347016045.19904.64.camel@luna.madrid.commsmundi.com>
	<5049E4B2.5090205@puzzled.xs4all.nl>
	<1347024299.19904.72.camel@luna.madrid.commsmundi.com>
Message-ID: <504A072C.2000101@puzzled.xs4all.nl>

On 09/07/2012 03:24 PM, Fran?ois Delawarde wrote:
> Hi Patrick,
>
> Not sure I understand the question. Aren't all LCD screens backlit?

The Cisco 7961 in my home office does not have a backlit LCD screen 
(like the Polycom IP670). Quite annoying in low light conditions so it's 
gathering dust.

> But yes the screen is a 320x160 LCD with backlight. About keys, you have
> 6 programmable keys on the top part and 10 more on the right with LEDs
> for BLF or other things. The other keys (dtmf, navigation, volume, etc.)
> have no light.

Thanks for the info.

Regards,
Patrick



From sdame at 207me.com  Fri Sep  7 18:39:01 2012
From: sdame at 207me.com (Stephen Dame)
Date: Fri, 7 Sep 2012 10:39:01 -0400
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <CAAyX+KFL-vjBODeqVG190wxenNu=gPjDoZOvJ=TOmu8YfzS84w@mail.gmail.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
	<CAAyX+KFL-vjBODeqVG190wxenNu=gPjDoZOvJ=TOmu8YfzS84w@mail.gmail.com>
Message-ID: <006e01cd8d06$86234130$9269c390$@207me.com>

Brett,

 

For simple stuff that doesn't need to scale, I use a 5th approach,   I just
use bash scripts, php, and a few ajax calls that just call fs_cli -X. to do
100% of functionality.   built a simple switchboard that shows live events,
plays sound cues, transfers calls into conferences etc.   ajax is getting
events every second to update UI.

 

Not pretty, but just took a few days to code and get into production.

 

Regards,

Stephen

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ben
Langfeld
Sent: Friday, September 07, 2012 10:01 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] best FS interface for development

 

The Adhearsion framework firmly takes the 3PCC approach via inbound event
socket, and does a similar thing on Asterisk. We believe that to be the best
way to isolate business logic from the switch.


Regards,
Ben Langfeld



On 7 September 2012 15:43, Brett Clark - Grasshopper
<bclark at grasshopper.com> wrote:

There seems to be at least 4 general ways for customizing FS:

- Use the event socket and build an app to interface with it

- Use mod lua, or similar, to develop in a particular language

- Develop your own 'module' to interface directly with the core

- Use the XML interface to implement IVR, PBX, and voicemail type
functionality and all the rest.

 

How prevalent is each approach in the community?  It seems like everyone is
doing a little of both, but maybe someone has a better handle of what the
actual ratios are?  

 

Which is best suited for an arbitrarily large and complex application?  I
realize this isn't a simple question-what I want is to understand the most
evolved and featureful way to interface with FS?  I don't want to adopt an
approach which isn't being actively maintained or is missing features.  As
new stuff is added to FS, which approach will allow me to adopt those new
features most easily?

 

Thanks!
Brett

 

 

 

 

 


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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Sep  7 18:42:08 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 7 Sep 2012 11:42:08 -0300
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <CAAyX+KFL-vjBODeqVG190wxenNu=gPjDoZOvJ=TOmu8YfzS84w@mail.gmail.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
	<CAAyX+KFL-vjBODeqVG190wxenNu=gPjDoZOvJ=TOmu8YfzS84w@mail.gmail.com>
Message-ID: <CABH8QX0Ysre+74PfbC=bx6sqmbiZ37J=N-QXXgGoA_0EeRW_aQ@mail.gmail.com>

I guess it all depends on what exactly you are trying to do.

I would say that XML cURL is by far the easiest approach if you are
handling simple scenarios that don't need in call interaction. ESL is the
most powerful but also the most complex one. Hope that helps.

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions



On Fri, Sep 7, 2012 at 11:00 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> The Adhearsion framework firmly takes the 3PCC approach via inbound event
> socket, and does a similar thing on Asterisk. We believe that to be the
> best way to isolate business logic from the switch.
>
> Regards,
> Ben Langfeld
>
>
> On 7 September 2012 15:43, Brett Clark - Grasshopper <
> bclark at grasshopper.com> wrote:
>
>> There seems to be at least 4 general ways for customizing FS:****
>>
>> - Use the event socket and build an app to interface with it****
>>
>> - Use mod lua, or similar, to develop in a particular language****
>>
>> - Develop your own ?module? to interface directly with the core****
>>
>> - Use the XML interface to implement IVR, PBX, and voicemail type
>> functionality and all the rest.****
>>
>> ** **
>>
>> How prevalent is each approach in the community?  It seems like everyone
>> is doing a little of both, but maybe someone has a better handle of what
>> the actual ratios are?  ****
>>
>> ** **
>>
>> Which is best suited for an arbitrarily large and complex application?  I
>> realize this isn?t a simple question?what I want is to understand the most
>> evolved and featureful way to interface with FS?  I don?t want to adopt an
>> approach which isn?t being actively maintained or is missing features.  As
>> new stuff is added to FS, which approach will allow me to adopt those new
>> features most easily?****
>>
>> ** **
>>
>> Thanks!
>> Brett****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Sep  7 18:47:50 2012
From: vishal.kakkar at gmail.com (Vishal Kakkar)
Date: Fri, 7 Sep 2012 20:17:50 +0530
Subject: [Freeswitch-users] Freeswitch compatible GSM Gateway for 30+
	channels
Message-ID: <CAAxwcqtdzzPM0jKrrfr0bKTn-pEHNzh2B7z2oqSCH=eVB3dWeQ@mail.gmail.com>

Hi All,

I am looking for *Freeswitch compatible GSM Gateway for 30+ channels(SIMS)*.
Please suggest in case you have used any.

Thanks.
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From matt.putnam at lightspar.com  Fri Sep  7 18:50:45 2012
From: matt.putnam at lightspar.com (Matt Putnam)
Date: Fri, 7 Sep 2012 09:50:45 -0500
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <CAKzWOxVOqtNU6Y6VBR-QYgu2Fy6U3XGD1Xiw3yUkq-c2Wj2qCQ@mail.gmail.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
	<CAKzWOxVOqtNU6Y6VBR-QYgu2Fy6U3XGD1Xiw3yUkq-c2Wj2qCQ@mail.gmail.com>
Message-ID: <9757304EEA8CE74494B6D21E63DA005212AE44D322@MBX23.exg5.exghost.com>

Thanks for the response Michael. Below is my code that I have been using in my directory the problem is that when the call is placed to the user they see the call as coming from lightspar1 not 9725551212. Which to me would mean I would have to have a registration for each DID that would be routed to a box with registration which I would like to avoid. What I am trying to find is a way to pass the destination number received on the gateway through to the registered user instead of it using the username as the destination number. I did find in the documentation for sofia where it mentions appending a ^<toheader> but that doesn't appear to work with the user/ it just returns a warning that user not found.

Gateway Config
<extension name="9725551212">
   <condition field="destination_number" expression="^9725551212$">
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge" data="user/lightspar1@${domain}"/>
    <action application="hangup"/>
   </condition>
</extension>

PBX Output:
Dialplan: sofia/external/9725551213 at 192.168.1.2 Regex (FAIL) [9725551212] destination_number(lightspar1) =~ /^9725551212$/ break=on-false

Thanks,
Matt Putnam
matt.putnam at lightspar.com

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, September 06, 2012 5:35 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch

No worries - it is indeed a total paradigm shift and things may seem unusual. If you have a PBX registered with FreeSWITCH then from the FS perspective it's a "user". Don't let the name "user" fool you - it's just a label, and it's easier to write than "SIP registered endpoint."

So, if your trunk is registered then that means you've got an entry in conf/directory/default/ that defines the "user". An example in there already is 1000.xml, where the id="1000". So to route a call to that "user" just do this:

<action application="bridge" data="user/1000"/>

That's it! FS does a lot of magic behind the scenes. So in the case of your DID, you need to add a dialplan entry. I recommend making a copy of conf/dialplan/default/00_inbound_did.xml and editing it to suit your needs. Let's say that your DID is 8005551212<tel:8005551212> and that your PBX is registered as user 1234. This dialplan entry would route an inbound DID call to your PBX:

<extension name="DID">
  <condition field="destination_number" expression="^\+?1?8005551212<tel:8005551212>">
    <action application="bridge" data="user/1234"/>
  </condition>
</extension>

Note I added some regex magic to strip out optional leading + or 1.

Let us know how that works. Be sure to join IRC if you want to discuss it in real time.
-MC
On Thu, Sep 6, 2012 at 12:09 PM, Matt Putnam <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
Sorry for the confusion I can really only equate it to asterisk as that's what our current platform is. Essentially the problem is that if I have a PBX that has a trunk that is registered to freeswitch and a call is received for a DID  that is on that trunk I am currently unable to send that DID to the PBX. When the call is received on the PBX side it looks as if the call is for the trunk instead of the DID is should be for. So in the SIP invite instead of using lightspar1 at domain it would use 5551212 at domain. In asterisk this was simply accomplished by a dial statement of (SIP/5551212 at lightspar1) what I am looking for is that equivalent in freeswitch.

Thanks,
Matt Putnam
matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>

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 Brian Foster
Sent: Thursday, September 06, 2012 10:48 AM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch


I'm confused. I've read your message about three times, still don't completely understand.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 6, 2012 10:44 AM, "Matt Putnam" <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
Thanks for the Response Mike I am able to get the trunk registered the real issue I guess is passing calls to that trunk. As an example I have freeswitch1 for customers which has a trunk lightspar1 registered to my gateway freeswitch box. When a call is placed to a DID that is associated to freeswitch1 I can see the call in the logs but the destination number is lightspar1 not the original DID that was called. Is there a way to set the from field to use the DID called instead of the trunk name so instead of lightspar1 at blah it would send the invite with NPANXXXXXX at blah?

Thanks,
Matt Putnam
matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Wednesday, September 05, 2012 6:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch

Hi Matt,

Welcome to FreeSWITCH!

If you want something to register with FreeSWITCH then simply add an entry in the user directory. Whether it's a "user" or a "trunk" really doesn't matter - it's just a SIP registration. Look in conf/directory/default/1000.xml (if using the example "vanilla" configuration) for a taste of what you need. Let us know if you have any other questions or join us in #freeswitch on irc.freenode.net<http://irc.freenode.net>.

-MC (IRC:mercutioviz)
On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com<mailto:matt.putnam at lightspar.com>> wrote:
This may have been asked before but my research hasn't turned up any results. I am currently testing freeswitch as a replacement to our current asterisk gateways. I think I have the basics of ip based trunks down but I am running into an issue with trunks registering to the freeswitch box. In asterisk it was a fairly simple process of giving the trunk a username and password and setting it to dynamic to get it to register. What would be the equivalent in freeswitch that would allow me to have customer trunks register to the system?


Thanks,
Matt Putnam


_________________________________________________________________________
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
________________________________
No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12

_________________________________________________________________________
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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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: 2437/5252 - Release Date: 09/06/12

_________________________________________________________________________
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

________________________________
No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 2012.0.2197 / Virus Database: 2437/5252 - Release Date: 09/06/12
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From avi at avimarcus.net  Fri Sep  7 19:02:27 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 7 Sep 2012 18:02:27 +0300
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <006e01cd8d06$86234130$9269c390$@207me.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
	<CAAyX+KFL-vjBODeqVG190wxenNu=gPjDoZOvJ=TOmu8YfzS84w@mail.gmail.com>
	<006e01cd8d06$86234130$9269c390$@207me.com>
Message-ID: <CALNZuwmvHTc01QBKV1ER2C+bA04ZdOk22r8=MZ9Sv0VWyRkM7Q@mail.gmail.com>

Stephen, I presume you have an XML dialplan for basic call routing? Brett's
question seemed to encompass ALL configuration, not just viewing live
events.

-Avi


On Fri, Sep 7, 2012 at 5:39 PM, Stephen Dame <sdame at 207me.com> wrote:

> Brett,****
>
> ** **
>
> For simple stuff that doesn?t need to scale, I use a 5th approach,   I
> just use bash scripts, php, and a few ajax calls that just call fs_cli ?X.
> to do 100% of functionality.   built a simple switchboard that shows live
> events, plays sound cues, transfers calls into conferences etc.   ajax is
> getting events every second to update UI.****
>
> ** **
>
> Not pretty, but just took a few days to code and get into production.****
>
> ** **
>
> Regards,****
>
> Stephen****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ben Langfeld
> *Sent:* Friday, September 07, 2012 10:01 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] best FS interface for development****
>
> ** **
>
> The Adhearsion framework firmly takes the 3PCC approach via inbound event
> socket, and does a similar thing on Asterisk. We believe that to be the
> best way to isolate business logic from the switch.****
>
>
> Regards,
> Ben Langfeld
>
> ****
>
> On 7 September 2012 15:43, Brett Clark - Grasshopper <
> bclark at grasshopper.com> wrote:****
>
> There seems to be at least 4 general ways for customizing FS:****
>
> - Use the event socket and build an app to interface with it****
>
> - Use mod lua, or similar, to develop in a particular language****
>
> - Develop your own ?module? to interface directly with the core****
>
> - Use the XML interface to implement IVR, PBX, and voicemail type
> functionality and all the rest.****
>
>  ****
>
> How prevalent is each approach in the community?  It seems like everyone
> is doing a little of both, but maybe someone has a better handle of what
> the actual ratios are?  ****
>
>  ****
>
> Which is best suited for an arbitrarily large and complex application?  I
> realize this isn?t a simple question?what I want is to understand the most
> evolved and featureful way to interface with FS?  I don?t want to adopt an
> approach which isn?t being actively maintained or is missing features.  As
> new stuff is added to FS, which approach will allow me to adopt those new
> features most easily?****
>
>  ****
>
> Thanks!
> Brett****
>
>  ****
>
>  ****
>
>  ****
>
>  ****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep  7 19:09:36 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 7 Sep 2012 12:09:36 -0300
Subject: [Freeswitch-users] Freeswitch compatible GSM Gateway for 30+
	channels
In-Reply-To: <CAAxwcqtdzzPM0jKrrfr0bKTn-pEHNzh2B7z2oqSCH=eVB3dWeQ@mail.gmail.com>
References: <CAAxwcqtdzzPM0jKrrfr0bKTn-pEHNzh2B7z2oqSCH=eVB3dWeQ@mail.gmail.com>
Message-ID: <CABH8QX0wDxz8VCA4MsJoAASsP-pts=+tTUrfmEH0==qmYNaW3Q@mail.gmail.com>

Khomp has boards that are compatible and they work very well. It can scale
because it has 16 channels per board and the board does not use a PCI or
PCI-Express slot. They have something called EBS (External Board Series)
that uses a regular ethernet cable to communicate with the server over a
proprietary protocol.

Regards,
Jo?o Mesquita



On Fri, Sep 7, 2012 at 11:47 AM, Vishal Kakkar <vishal.kakkar at gmail.com>wrote:

> Hi All,
>
> I am looking for *Freeswitch compatible GSM Gateway for 30+ channels(SIMS)
> *.
> Please suggest in case you have used any.
>
> 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 paul at cupis.co.uk  Fri Sep  7 20:03:35 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Fri, 7 Sep 2012 17:03:35 +0100
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44D322@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
	<CAKzWOxVOqtNU6Y6VBR-QYgu2Fy6U3XGD1Xiw3yUkq-c2Wj2qCQ@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D322@MBX23.exg5.exghost.com>
Message-ID: <20120907160335.GA16394@eagle.cupis.co.uk>

On Fri, Sep 07, 2012 at 09:50:45AM -0500, Matt Putnam wrote:
>    Thanks for the response Michael. Below is my code that I have been
>    using in my directory the problem is that when the call is placed to
>    the user they see the call as coming from lightspar1 not 9725551212.

I'm not suggesting that this is necessarily the "correct" way to
implement what you are trying to setup, but you may find that this works
for this specific scenario:

    <action application="bridge" data="{sip_invite_req_uri=sip:${destination_number}@${domain}}user/lightspar1@${domain}"/>

Regards,



From freeswitch at scottisheyes.com  Fri Sep  7 20:11:06 2012
From: freeswitch at scottisheyes.com (James)
Date: Fri, 7 Sep 2012 09:11:06 -0700
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44D322@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
	<CAKzWOxVOqtNU6Y6VBR-QYgu2Fy6U3XGD1Xiw3yUkq-c2Wj2qCQ@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D322@MBX23.exg5.exghost.com>
Message-ID: <CALfx8k3s3XXFrMei1zebBtKzG5Try1-_YrAnECN61vzK8mm5Kg@mail.gmail.com>

>From your PBX output you listed...  I'm no expert here, but the regex won't
match the incoming number: assuming that your domain is
actually "192.168.1.2", 972555121*3* won't match 972555121*2*.  But then,
I'm a novice, so perhaps I'm missing something.

Cheers,
James

On Fri, Sep 7, 2012 at 7:50 AM, Matt Putnam <matt.putnam at lightspar.com>wrote:

> Thanks for the response Michael. Below is my code that I have been using
> in my directory the problem is that when the call is placed to the user
> they see the call as coming from lightspar1 not 9725551212. Which to me
> would mean I would have to have a registration for each DID that would be
> routed to a box with registration which I would like to avoid. What I am
> trying to find is a way to pass the destination number received on the
> gateway through to the registered user instead of it using the username as
> the destination number. I did find in the documentation for sofia where it
> mentions appending a ^<toheader> but that doesn?t appear to work with the
> user/ it just returns a warning that user not found.  ****
>
> ** **
>
> Gateway Config****
>
> <extension name="9725551212">****
>
>    <condition field="destination_number" expression="^9725551212$">****
>
>     <action application="set" data="hangup_after_bridge=true"/>****
>
>     <action application="bridge" data="user/lightspar1@${domain}"/>****
>
>     <action application="hangup"/>****
>
>    </condition>****
>
> </extension>****
>
> ** **
>
> PBX Output:****
>
> Dialplan: sofia/external/9725551213 at 192.168.1.2 Regex (FAIL) [9725551212]
> destination_number(lightspar1) =~ /^9725551212$/ break=on-false****
>
> ** **
>
> Thanks,****
>
> Matt Putnam****
>
> matt.putnam at lightspar.com****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, September 06, 2012 5:35 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch****
>
> ** **
>
> No worries - it is indeed a total paradigm shift and things may seem
> unusual. If you have a PBX registered with FreeSWITCH then from the FS
> perspective it's a "user". Don't let the name "user" fool you - it's just a
> label, and it's easier to write than "SIP registered endpoint."
>
> So, if your trunk is registered then that means you've got an entry in
> conf/directory/default/ that defines the "user". An example in there
> already is 1000.xml, where the id="1000". So to route a call to that "user"
> just do this:
>
> <action application="bridge" data="user/1000"/>
>
> That's it! FS does a lot of magic behind the scenes. So in the case of
> your DID, you need to add a dialplan entry. I recommend making a copy of
> conf/dialplan/default/00_inbound_did.xml and editing it to suit your needs.
> Let's say that your DID is 8005551212 and that your PBX is registered as
> user 1234. This dialplan entry would route an inbound DID call to your PBX:
>
> <extension name="DID">
>   <condition field="destination_number" expression="^\+?1?8005551212">
>     <action application="bridge" data="user/1234"/>
>   </condition>
> </extension>
>
> Note I added some regex magic to strip out optional leading + or 1.
>
> Let us know how that works. Be sure to join IRC if you want to discuss it
> in real time.
> -MC****
>
> On Thu, Sep 6, 2012 at 12:09 PM, Matt Putnam <matt.putnam at lightspar.com>
> wrote:****
>
> Sorry for the confusion I can really only equate it to asterisk as that?s
> what our current platform is. Essentially the problem is that if I have a
> PBX that has a trunk that is registered to freeswitch and a call is
> received for a DID  that is on that trunk I am currently unable to send
> that DID to the PBX. When the call is received on the PBX side it looks as
> if the call is for the trunk instead of the DID is should be for. So in the
> SIP invite instead of using lightspar1 at domain it would use 5551212 at domain.In asterisk this was simply accomplished by a dial statement of
> (SIP/5551212 at lightspar1) what I am looking for is that equivalent in
> freeswitch.****
>
>  ****
>
> Thanks,****
>
> Matt Putnam****
>
> matt.putnam at lightspar.com****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, September 06, 2012 10:48 AM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch****
>
>  ****
>
> I'm confused. I've read your message about three times, still don't
> completely understand.****
>
> Brian Foster
> Endigo Computer LLC****
>
> Sent from a mobile device.****
>
> On Sep 6, 2012 10:44 AM, "Matt Putnam" <matt.putnam at lightspar.com> wrote:*
> ***
>
> Thanks for the Response Mike I am able to get the trunk registered the
> real issue I guess is passing calls to that trunk. As an example I have
> freeswitch1 for customers which has a trunk lightspar1 registered to my
> gateway freeswitch box. When a call is placed to a DID that is associated
> to freeswitch1 I can see the call in the logs but the destination number is
> lightspar1 not the original DID that was called. Is there a way to set the
> from field to use the DID called instead of the trunk name so instead of
> lightspar1 at blah it would send the invite with NPANXXXXXX at blah?****
>
>  ****
>
> Thanks,****
>
> Matt Putnam****
>
> matt.putnam at lightspar.com****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, September 05, 2012 6:49 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch****
>
>  ****
>
> Hi Matt,
>
> Welcome to FreeSWITCH!
>
> If you want something to register with FreeSWITCH then simply add an entry
> in the user directory. Whether it's a "user" or a "trunk" really doesn't
> matter - it's just a SIP registration. Look in
> conf/directory/default/1000.xml (if using the example "vanilla"
> configuration) for a taste of what you need. Let us know if you have any
> other questions or join us in #freeswitch on irc.freenode.net.
>
> -MC (IRC:mercutioviz)****
>
> On Wed, Sep 5, 2012 at 3:21 PM, Matt Putnam <matt.putnam at lightspar.com>
> wrote:****
>
> This may have been asked before but my research hasn?t turned up any
> results. I am currently testing freeswitch as a replacement to our current
> asterisk gateways. I think I have the basics of ip based trunks down but I
> am running into an issue with trunks registering to the freeswitch box. In
> asterisk it was a fairly simple process of giving the trunk a username and
> password and setting it to dynamic to get it to register. What would be the
> equivalent in freeswitch that would allow me to have customer trunks
> register to the system? ****
>
>  ****
>
>  ****
>
> Thanks,****
>
> Matt Putnam****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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****
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2197 / Virus Database: 2437/5251 - Release Date: 09/05/12*
> ***
>
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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: 2012.0.2197 / Virus Database: 2437/5252 - Release Date: 09/06/12*
> ***
>
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> ****
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2197 / Virus Database: 2437/5252 - Release Date: 09/06/12*
> ***
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmrienzo at gmail.com  Fri Sep  7 20:13:59 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 7 Sep 2012 12:13:59 -0400
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
Message-ID: <CAD-qRkjgK9rGFmznjVxM-CiCFFgYKBSTwOR5RwHevBAfaaviQA@mail.gmail.com>

For arbitrarily large and complex systems, I'd keep FreeSWITCH as dumb as
possible so that you can easily cluster them.  So, modules that allow
external control (mod_event_socket, mod_erlang_event, mod_httapi,
mod_xml_curl) are all good choices.

Event socket is pretty popular; Plivo and Adhearsion both use it.  The
2600hz guys use mod_erlang_event.  And I've heard of plenty of people that
use mod_xml_curl.

Chris



On Fri, Sep 7, 2012 at 9:43 AM, Brett Clark - Grasshopper <
bclark at grasshopper.com> wrote:

> There seems to be at least 4 general ways for customizing FS:****
>
> - Use the event socket and build an app to interface with it****
>
> - Use mod lua, or similar, to develop in a particular language****
>
> - Develop your own ?module? to interface directly with the core****
>
> - Use the XML interface to implement IVR, PBX, and voicemail type
> functionality and all the rest.****
>
> ** **
>
> How prevalent is each approach in the community?  It seems like everyone
> is doing a little of both, but maybe someone has a better handle of what
> the actual ratios are?  ****
>
> ** **
>
> Which is best suited for an arbitrarily large and complex application?  I
> realize this isn?t a simple question?what I want is to understand the most
> evolved and featureful way to interface with FS?  I don?t want to adopt an
> approach which isn?t being actively maintained or is missing features.  As
> new stuff is added to FS, which approach will allow me to adopt those new
> features most easily?****
>
> ** **
>
> Thanks!
> Brett****
>
> ** **
>
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From carl-johan at seacom.dk  Fri Sep  7 15:27:55 2012
From: carl-johan at seacom.dk (Carl Johan Jensen)
Date: Fri, 7 Sep 2012 13:27:55 +0200
Subject: [Freeswitch-users] Numerous compile errors
Message-ID: <007c01cd8ceb$d39939d0$7acbad70$@dk>

Hello out there,

 

I have installed Visual C++ 2008 Express edition on my Windows 7 machine.  I
downloads the  freeswitch-1-0-6-tar-gz file from
http://files.freeswitch.org. 

I extract the source code and opens the project in Visual C++. Then I press
F7 to start compilation. Numerous errors are returned:

At the end of compilation:

120>LINK : fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib'

120>Build log was saved at
"file://c:\R&D\FreeSwitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\Win32
\Debug\BuildLog.htm"

120>mod_sofia - 1 error(s), 12 warning(s)

========== Rebuild All: 102 succeeded, 17 failed, 11 skipped ==========

 

I investigate the errors and find among several the below errors:

 

13>c1 : fatal error C1083: Cannot open source file:
'..\..\pthreads-w32-2-7-0-release\pthread.c': No such file or directory

13>Build log was saved at
"file://c:\R&D\FreeSwitch\freeswitch-1.0.6\libs\win32\pthread\Debug
DLL\BuildLog.htm"

13>pthread - 1 error(s), 0 warning(s)

 

14>Deleting intermediate and output files for project 'make_modem_filter',
configuration 'All|Win32'

14>make_modem_filter : error PRJ0008 : Could not delete file
'c:\R&D\FreeSwitch\freeswitch-1.0.6\libs\spandsp\src\msvc\All\RSP0001D534166
324.rsp'.

14>Make sure that the file is not open by another process and is not
write-protected.

14>make_modem_filter - 1 error(s), 0 warning(s)

 

19>agc.c

19>c1 : fatal error C1083: Cannot open source file:
'..\..\sphinxbase-0.4.99\src\libsphinxbase\feat\agc.c': No such file or
directory

19>bio.c

 

Anyone knows what I miss to do. (I am following the instructions of the book
"FreeSWITCH-106-eBook06092012_1187154")

 

 

Best Regards

Carl Johan 

 

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From hard_patel09 at yahoo.com  Fri Sep  7 19:38:01 2012
From: hard_patel09 at yahoo.com (Hardik Patel)
Date: Fri, 7 Sep 2012 08:38:01 -0700 (PDT)
Subject: [Freeswitch-users] mod_perl installation error
Message-ID: <1347032281.54902.YahooMailNeo@web111611.mail.gq1.yahoo.com>

Hi,

I am doing installation of freeswitch with mod_perl module but during compilation i got below error.

********ERROR**************
making all mod_perl
Creating mod_perl.la...
/usr/bin/ld: /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a(op.o): relocation R_X86_64_32S against `PL_sv_yes' can not be used when making a shared object; recompile with -fPIC
/usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a: could not read symbols: Bad value
collect2: ld returned 1 exit status
cat: .libs/mod_perl.log: No such file or directory
make[5]: *** [mod_perl.la] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
***********************************************

making all mod_perl
Creating mod_perl.la...
/usr/bin/ld: /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a(op.o): relocation R_X86_64_32S against `PL_sv_yes' can not be used when making a shared object; recompile with -fPIC
/usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a: could not read symbols: Bad value
collect2: ld returned 1 exit status
cat: .libs/mod_perl.log: No such file or directory
make[5]: *** [mod_perl.la] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

can anybody please help me to figure out the issue?

Thanks in advance.

Thanks,
Hardik Patel
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From philq at qsystemsengineering.com  Fri Sep  7 20:18:51 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Fri, 7 Sep 2012 09:18:51 -0700 (PDT)
Subject: [Freeswitch-users] More information on FreeSwitch
In-Reply-To: <CALNZuwnv1SzQ8FNC47U7SZuwAFo7u6JK3mxaGTmmOkVOfMF=rQ@mail.gmail.com>
References: <CC20783F-329B-4377-914E-CE4D640529C6@gmail.com>
	<CACc3D0XO5kOZ_imMy6XazzTGyYUpgJtrRRoAtkK9_WKTbFFo+w@mail.gmail.com>
	<CALNZuwnv1SzQ8FNC47U7SZuwAFo7u6JK3mxaGTmmOkVOfMF=rQ@mail.gmail.com>
Message-ID: <1347034731293-7582678.post@n2.nabble.com>

You may want to take a serious look at FusionPBX, it makes many of the more
mundane administrative and monitoring tasks a lot easier, especially if
you're not working with it and making changes every day.  It avoids much of
the "ok... how did I do that again?"  It's a nice tool for learning the
architecture of FS as you're getting started, although relying upon it too
heavily will likely slow your learning over the long haul.

You won't ever completely get away from editing files though, often vi is
the best administrative tool available.

I bought the original FreeSwitch book in addition to the cookbook and while
it is an excellent text with a lot of good information contained therein,
there's still some info that's not in there that I'd like to see.  For
instance, I'm still left with questions about FS' capabilities and behavior
with NAT handling in certain scenarios and that info doesn't appear to be
very well covered anywhere that I've been able to find - probably due to the
fact that FS is still a work in progress.  Not many people seem to truly
understand the details of FS' NAT functionality, so unless you're lucky and
one of the developers who truly understands it has the time to answer your
question, you'll have to wade through the code and experiment (which isn't
always a bad thing, another one of the many advantages of open source) but
that can be quite time-consuming.

I was surprised to find that the full original FS book is floating around
the 'Net in PDF form, not sure that the authors or the publisher would be
particularly happy about that.  If any of the AUTHORS want the link(s),
contact me and I'll give the info I have to you.  If your name doesn't
appear on the cover of the FS book however, don't ask me for any links.   :)

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-information-on-FreeSwitch-tp7582648p7582678.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Fri Sep  7 20:27:55 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 7 Sep 2012 09:27:55 -0700
Subject: [Freeswitch-users] Numerous compile errors
In-Reply-To: <007c01cd8ceb$d39939d0$7acbad70$@dk>
References: <007c01cd8ceb$d39939d0$7acbad70$@dk>
Message-ID: <CAMWiMy_DyZhF61aPkm4_JRu6sFuE0HZOwRRsJZ3ukyarAwtuFQ@mail.gmail.com>

I'm guessing that Collins had something to do with this one.

In the meantime, please use the 1.2.x tarballs. 1.0.6 isn't supported anymore.

On Fri, Sep 7, 2012 at 4:27 AM, Carl Johan Jensen <carl-johan at seacom.dk> wrote:
> Hello out there,
>
>
>
> I have installed Visual C++ 2008 Express edition on my Windows 7 machine.  I
> downloads the  freeswitch-1-0-6-tar-gz file from
> http://files.freeswitch.org.
>
> I extract the source code and opens the project in Visual C++. Then I press
> F7 to start compilation. Numerous errors are returned:
>
> At the end of compilation:
>
> 120>LINK : fatal error LNK1181: cannot open input file
> '..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib'
>
> 120>Build log was saved at
> "file://c:\R&D\FreeSwitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\Win32\Debug\BuildLog.htm"
>
> 120>mod_sofia - 1 error(s), 12 warning(s)
>
> ========== Rebuild All: 102 succeeded, 17 failed, 11 skipped ==========
>
>
>
> I investigate the errors and find among several the below errors:
>
>
>
> 13>c1 : fatal error C1083: Cannot open source file:
> '..\..\pthreads-w32-2-7-0-release\pthread.c': No such file or directory
>
> 13>Build log was saved at
> "file://c:\R&D\FreeSwitch\freeswitch-1.0.6\libs\win32\pthread\Debug
> DLL\BuildLog.htm"
>
> 13>pthread - 1 error(s), 0 warning(s)
>
>
>
> 14>Deleting intermediate and output files for project 'make_modem_filter',
> configuration 'All|Win32'
>
> 14>make_modem_filter : error PRJ0008 : Could not delete file
> 'c:\R&D\FreeSwitch\freeswitch-1.0.6\libs\spandsp\src\msvc\All\RSP0001D534166324.rsp'.
>
> 14>Make sure that the file is not open by another process and is not
> write-protected.
>
> 14>make_modem_filter - 1 error(s), 0 warning(s)
>
>
>
> 19>agc.c
>
> 19>c1 : fatal error C1083: Cannot open source file:
> '..\..\sphinxbase-0.4.99\src\libsphinxbase\feat\agc.c': No such file or
> directory
>
> 19>bio.c
>
>
>
> Anyone knows what I miss to do. (I am following the instructions of the book
> ?FreeSWITCH-106-eBook06092012_1187154?)
>
>
>
>
>
> Best Regards
>
> Carl Johan
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From matt.putnam at lightspar.com  Fri Sep  7 20:52:06 2012
From: matt.putnam at lightspar.com (Matt Putnam)
Date: Fri, 7 Sep 2012 11:52:06 -0500
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <20120907160335.GA16394@eagle.cupis.co.uk>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
	<CAKzWOxVOqtNU6Y6VBR-QYgu2Fy6U3XGD1Xiw3yUkq-c2Wj2qCQ@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D322@MBX23.exg5.exghost.com>
	<20120907160335.GA16394@eagle.cupis.co.uk>
Message-ID: <9757304EEA8CE74494B6D21E63DA005212AE44D3CA@MBX23.exg5.exghost.com>

Using your code with the variables causes freeswitch on my gateway to crash. If I replace the variables with hard coded number and ip it doesn't crash but I get the error below. 

2012-09-07 11:37:41.036394 [NOTICE] switch_ivr_originate.c:2554 Cannot create outgoing channel of type [user] cause: [EXCHANGE_ROUT ING_ERROR]

Thanks,
Matt Putnam
matt.putnam at lightspar.com

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis
Sent: Friday, September 07, 2012 11:04 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Registering SIP Trunks to Freeswitch

On Fri, Sep 07, 2012 at 09:50:45AM -0500, Matt Putnam wrote:
>    Thanks for the response Michael. Below is my code that I have been
>    using in my directory the problem is that when the call is placed to
>    the user they see the call as coming from lightspar1 not 9725551212.

I'm not suggesting that this is necessarily the "correct" way to implement what you are trying to setup, but you may find that this works for this specific scenario:

    <action application="bridge" data="{sip_invite_req_uri=sip:${destination_number}@${domain}}user/lightspar1@${domain}"/>

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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: 2012.0.2197 / Virus Database: 2437/5254 - Release Date: 09/07/12


From mike.burlingame at me.com  Fri Sep  7 21:00:13 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 07 Sep 2012 10:00:13 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
Message-ID: <0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>

after putting the server in production and  loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.

However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.



On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame at me.com> wrote:

> After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
> 
> Thanks
> 
> On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> 
>> Looks much much better Thank you -- Now to conduct more testing 
>> 
>> 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-Leg:5060, with session description
>> 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying
>> 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>> 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>> 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
>> 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description
>> 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
>> 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
>> 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK
>> 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
>> 
>> 
>> 
>> On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> ok,
>>> 
>>> update one more time, if it still does not work just go right to jira
>>> with the latest (not before today's changes)
>>> 
>>> 
>>> On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
>>>> 
>>>> commit d45db898339e1b2212f5befff1af714abcec034f
>>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>> Date:   Wed Sep 5 13:11:32 2012 -0500
>>>> 
>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>>>> 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>> 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>>> 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>> 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>> 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
>>>> 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>>>> 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>> 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>> 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>>>> 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>>>> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>>> 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>>> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
>>>> 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
>>>> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
>>>> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>>>> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>>>> 
>>>> 
>>>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>> 
>>>>> update and try again, if it still doesn't work open a jira
>>>>> 
>>>>> 
>>>>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>>> as promised here is the update testing and enabling
>>>>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
>>>>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
>>>>>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>>>>>> 
>>>>>> I can open a JIRA case on this and provide the console log file / PCAP's ect
>>>>>> if that would help
>>>>>> 
>>>>>> 
>>>>>> Call Flow with out {sip_wait_for_aleg_ack=true}
>>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>>>>> sip:+13605551212 at A-LEG:5060, with session description
>>>>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>>>>> sip:13605551212 at B-Leg, with session description
>>>>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
>>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
>>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>>>>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>>>> 
>>>>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>>>>> sip:+13605551212 at A-LEG:5060, with session description
>>>>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>>>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>>>>> sip:13605551212 at B-Leg, with session description
>>>>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>>>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>>>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>>>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>>>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>>>>> description
>>>>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
>>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
>>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>>>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>>>>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>>>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>>>>>> 
>>>>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>>> 
>>>>>> No worries I will be out this weekend for the long weekend I will work on
>>>>>> getting the test box upgraded and a test case setup on Tuesday I will report
>>>>>> back the results mid to late next week and provided everything works as I
>>>>>> hope it will I will be happy to pay the Wiki tax :)
>>>>>> 
>>>>>> 
>>>>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
>>>>>> wrote:
>>>>>>> 
>>>>>>> Cool I will nail that up on my test box and see if that works
>>>>>>> 
>>>>>> Please report back on whether it works or not and then be prepared to pay
>>>>>> the wiki tax. :) I'll be glad to assist with getting this documented
>>>>>> although I think you're in the best position to give that documentation some
>>>>>> real-world context.
>>>>>> 
>>>>>> -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
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Sep  7 21:16:12 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Sep 2012 12:16:12 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
Message-ID: <CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>

FYI: You need to get the backtrace before you update or the core file
becomes useless.
It dumps core by default so you should have one but if you updated, its
tool late unless you can checkout the exact previous rev.



On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> after putting the server in production and  loading it up it just randomly
> crashed I am updating to the latest git will try to capture a core dump if
> one is created on exit nothing in the logs point to anything it was
> handling calls and just exited.
>
> However on a positive note the switch sip_wait_for_aleg_ack=true seems to
> do as expected with all my tests.
>
>
>
> On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame at me.com>
> wrote:
>
> > After about 20K test calls this seems to have addressed the issue - I
> will keep running my test's for today and put this box in a production
> environment tomorrow to validate it still holds up with load. I will report
> back after that is completed.
> >
> > Thanks
> >
> > On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame at me.com>
> wrote:
> >
> >> Looks much much better Thank you -- Now to conduct more testing
> >>
> >> 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-Leg:5060,
> with session description
> >> 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying
> >> 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg,
> with session description
> >> 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >> 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >> 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
> >> 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >> 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >> 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session
> description
> >> 5.385087 A-Leg -> FreeSwitch SIP Request: ACK
> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >> 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >> 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212
> @FreeSwitch:5070;transport=udp
> >> 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK
> >> 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >> 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>
> >>
> >>
> >> On Sep 5, 2012, at 12:01 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
> >>
> >>> ok,
> >>>
> >>> update one more time, if it still does not work just go right to jira
> >>> with the latest (not before today's changes)
> >>>
> >>>
> >>> On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <
> mike.burlingame at me.com> wrote:
> >>>> The change seems to have broken the ability for the call to connect -
> would you like me to open a jira up with the current log files or before
> the change was made today?
> >>>>
> >>>> commit d45db898339e1b2212f5befff1af714abcec034f
> >>>> Author: Anthony Minessale <anthm at freeswitch.org>
> >>>> Date:   Wed Sep 5 13:11:32 2012 -0500
> >>>>
> >>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060,
> with session description
> >>>> 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>> 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg,
> with session description
> >>>> 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>> 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>> 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
> >>>> 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG
> :5060
> >>>> 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>> 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>> 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction
> does not exist
> >>>> 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:
> +13605551212 at A-LEG:5060, with session description
> >>>> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg,
> with session description
> >>>> 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> description
> >>>> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL
> sip:+13605551212 at A-LEG:5060
> >>>> 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
> >>>> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212
> @A-LEG:5060
> >>>> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction
> does not exist
> >>>>
> >>>>
> >>>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
> >>>>
> >>>>> update and try again, if it still doesn't work open a jira
> >>>>>
> >>>>>
> >>>>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <
> mike.burlingame at me.com> wrote:
> >>>>>> as promised here is the update testing and enabling
> >>>>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio
> starts to
> >>>>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS
> from the
> >>>>>> B-Leg as well as an abnormally long delay in getting an ACK from
> the A-Leg
> >>>>>>
> >>>>>> I can open a JIRA case on this and provide the console log file /
> PCAP's ect
> >>>>>> if that would help
> >>>>>>
> >>>>>>
> >>>>>> Call Flow with out {sip_wait_for_aleg_ack=true}
> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
> >>>>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >>>>>> sip:13605551212 at B-Leg, with session description
> >>>>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>>>>>
> >>>>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
> >>>>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >>>>>> sip:13605551212 at B-Leg, with session description
> >>>>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>>>>>
> >>>>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <
> mike.burlingame at me.com> wrote:
> >>>>>>
> >>>>>> No worries I will be out this weekend for the long weekend I will
> work on
> >>>>>> getting the test box upgraded and a test case setup on Tuesday I
> will report
> >>>>>> back the results mid to late next week and provided everything
> works as I
> >>>>>> hope it will I will be happy to pay the Wiki tax :)
> >>>>>>
> >>>>>>
> >>>>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <
> mike.burlingame at me.com>
> >>>>>> wrote:
> >>>>>>>
> >>>>>>> Cool I will nail that up on my test box and see if that works
> >>>>>>>
> >>>>>> Please report back on whether it works or not and then be prepared
> to pay
> >>>>>> the wiki tax. :) I'll be glad to assist with getting this documented
> >>>>>> although I think you're in the best position to give that
> documentation some
> >>>>>> real-world context.
> >>>>>>
> >>>>>> -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
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at 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
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike.burlingame at me.com  Fri Sep  7 21:23:31 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 07 Sep 2012 10:23:31 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
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	<C9A00194-61D5-4DB8-A5F6-0953F071CEAF@me.com>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
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	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
Message-ID: <C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>

I figured I would update to the latest git due to in the past I have seen "make sure your running the latest git" the odd thing is no core dump was created on the last crash.

I will post the backtrace if a core dump is generated after the next crash


On Sep 7, 2012, at 10:16 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> FYI: You need to get the backtrace before you update or the core file becomes useless.
> It dumps core by default so you should have one but if you updated, its tool late unless you can checkout the exact previous rev.
> 
> 
> 
> On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> after putting the server in production and  loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.
> 
> However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.
> 
> 
> 
> On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> 
> > After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
> >
> > Thanks
> >
> > On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> >
> >> Looks much much better Thank you -- Now to conduct more testing
> >>
> >> 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-Leg:5060, with session description
> >> 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying
> >> 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
> >> 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >> 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >> 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
> >> 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >> 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >> 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description
> >> 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
> >> 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >> 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
> >> 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK
> >> 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >> 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>
> >>
> >>
> >> On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >>
> >>> ok,
> >>>
> >>> update one more time, if it still does not work just go right to jira
> >>> with the latest (not before today's changes)
> >>>
> >>>
> >>> On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> >>>> The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
> >>>>
> >>>> commit d45db898339e1b2212f5befff1af714abcec034f
> >>>> Author: Anthony Minessale <anthm at freeswitch.org>
> >>>> Date:   Wed Sep 5 13:11:32 2012 -0500
> >>>>
> >>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
> >>>> 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>> 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
> >>>> 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>> 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>> 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
> >>>> 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
> >>>> 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>> 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>> 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
> >>>> 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
> >>>> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
> >>>> 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
> >>>> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
> >>>> 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
> >>>> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
> >>>> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
> >>>>
> >>>>
> >>>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >>>>
> >>>>> update and try again, if it still doesn't work open a jira
> >>>>>
> >>>>>
> >>>>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> >>>>>> as promised here is the update testing and enabling
> >>>>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
> >>>>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
> >>>>>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
> >>>>>>
> >>>>>> I can open a JIRA case on this and provide the console log file / PCAP's ect
> >>>>>> if that would help
> >>>>>>
> >>>>>>
> >>>>>> Call Flow with out {sip_wait_for_aleg_ack=true}
> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
> >>>>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >>>>>> sip:13605551212 at B-Leg, with session description
> >>>>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>>>>>
> >>>>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
> >>>>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
> >>>>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
> >>>>>> sip:13605551212 at B-Leg, with session description
> >>>>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
> >>>>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
> >>>>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
> >>>>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
> >>>>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
> >>>>>> description
> >>>>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
> >>>>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
> >>>>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
> >>>>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
> >>>>>>
> >>>>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> >>>>>>
> >>>>>> No worries I will be out this weekend for the long weekend I will work on
> >>>>>> getting the test box upgraded and a test case setup on Tuesday I will report
> >>>>>> back the results mid to late next week and provided everything works as I
> >>>>>> hope it will I will be happy to pay the Wiki tax :)
> >>>>>>
> >>>>>>
> >>>>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
> >>>>>> wrote:
> >>>>>>>
> >>>>>>> Cool I will nail that up on my test box and see if that works
> >>>>>>>
> >>>>>> Please report back on whether it works or not and then be prepared to pay
> >>>>>> the wiki tax. :) I'll be glad to assist with getting this documented
> >>>>>> although I think you're in the best position to give that documentation some
> >>>>>> real-world context.
> >>>>>>
> >>>>>> -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
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at 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
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sdame at 207me.com  Fri Sep  7 21:58:37 2012
From: sdame at 207me.com (Stephen Dame)
Date: Fri, 7 Sep 2012 13:58:37 -0400
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <CALNZuwmvHTc01QBKV1ER2C+bA04ZdOk22r8=MZ9Sv0VWyRkM7Q@mail.gmail.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com><CAAyX+KFL-vjBODeqVG190wxenNu=gPjDoZOvJ=TOmu8
	YfzS84w@mail.gmail.com><006e01cd8d06$86234130$9269c390$@207me.com>
	<CALNZuwmvHTc01QBKV1ER2C+bA04ZdOk22r8=MZ9Sv0VWyRkM7Q@mail.gmail.com>
Message-ID: <00e901cd8d22$6894d040$39be70c0$@207me.com>

Yes, my app is a live radio station switch board with freeswitch/icecast, so
the dialplan and configurations for conferences, skype, and DID inbound
calling are all statically configured in default xml files.

 

stephen

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Friday, September 07, 2012 11:02 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] best FS interface for development

 

Stephen, I presume you have an XML dialplan for basic call routing? Brett's
question seemed to encompass ALL configuration, not just viewing live
events.




-Avi

 

On Fri, Sep 7, 2012 at 5:39 PM, Stephen Dame <sdame at 207me.com> wrote:

Brett,

 

For simple stuff that doesn't need to scale, I use a 5th approach,   I just
use bash scripts, php, and a few ajax calls that just call fs_cli -X. to do
100% of functionality.   built a simple switchboard that shows live events,
plays sound cues, transfers calls into conferences etc.   ajax is getting
events every second to update UI.

 

Not pretty, but just took a few days to code and get into production.

 

Regards,

Stephen

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ben
Langfeld
Sent: Friday, September 07, 2012 10:01 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] best FS interface for development

 

The Adhearsion framework firmly takes the 3PCC approach via inbound event
socket, and does a similar thing on Asterisk. We believe that to be the best
way to isolate business logic from the switch.


Regards,
Ben Langfeld

On 7 September 2012 15:43, Brett Clark - Grasshopper
<bclark at grasshopper.com> wrote:

There seems to be at least 4 general ways for customizing FS:

- Use the event socket and build an app to interface with it

- Use mod lua, or similar, to develop in a particular language

- Develop your own 'module' to interface directly with the core

- Use the XML interface to implement IVR, PBX, and voicemail type
functionality and all the rest.

 

How prevalent is each approach in the community?  It seems like everyone is
doing a little of both, but maybe someone has a better handle of what the
actual ratios are?  

 

Which is best suited for an arbitrarily large and complex application?  I
realize this isn't a simple question-what I want is to understand the most
evolved and featureful way to interface with FS?  I don't want to adopt an
approach which isn't being actively maintained or is missing features.  As
new stuff is added to FS, which approach will allow me to adopt those new
features most easily?

 

Thanks!
Brett

 

 

 

 

 


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




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

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

 


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




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

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

 

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From bedgar at vseinc.com  Fri Sep  7 22:49:15 2012
From: bedgar at vseinc.com (bedgar at vseinc.com)
Date: Fri, 7 Sep 2012 14:49:15 -0400
Subject: [Freeswitch-users] freeswitch - sangoma CAS T1 -
	a101	configuration
In-Reply-To: <CAKan7GpQ_v=gFuho3b+i_EmyYg4sJ4-JB8JFt9XhMhgkh6t=qg@mail.gmail.com>
References: <CAKan7GpuY9zFG=BrPuONw5-eKOiK_+1tyOA6zE8GvwbwgG5k_A@mail.gmail.com>
	<CAMWiMy9n1rfC9euxT6_DPxJFJ80-xhoBsZQEjcR-QmX3CaCLKg@mail.gmail.com>
	<CAKan7Gows2HJZfVB+W9JmBp7jWmPgfG7sJvMeJZo1rvRNWHbRA@mail.gmail.com>
	<CAA4nhyAtt4yp6_VmAbYyi0bTQosGa1=mpSH_4LMdzj90YDQT3g@mail.gmail.com>
	<CAKan7GpQ_v=gFuho3b+i_EmyYg4sJ4-JB8JFt9XhMhgkh6t=qg@mail.gmail.com>
Message-ID: <333789DE5C38474EB3A478A538F4EBAB0A987CED40@prod-exch01.corp.vseinc.com>

Has this issue been resolved.  I am also having problems getting Sangoma A101D working on FS using and RBS T-1.  Getting one way audio, out but not in.

Brian C. Edgar, Jr.
Senior Systems Administrator
Voice Systems Engineering Inc.
Email: bedgar at vseinc.com<mailto:bedgar at vseinc.com>
Phone: 215-953-8568 x278

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of copycall
Sent: Monday, May 28, 2012 5:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch - sangoma CAS T1 - a101 configuration

moises,

thank you for your response.

the T1 linecard at the end of this call is CAS T1, with B8ZS, and ESF, extended super frame, with E&M trunk signaling.

as i understand it, this is not robbed bit signaling, but a later variation of it.

i will paste the sangoma linecard config, below:

########################################################################
#                       Sangoma Wanpipe                             #
#        Dahdi/Zaptel/SMG/TDMAPI/BOOT Configuration Script             #
#                             v2.39                                  #
#                     Sangoma Technologies Inc.                        #
#                        Copyright(c) 2009.                            #
########################################################################

Would you like to change FreeSWITCH Configuration Directory?
Default: /usr/local/freeswitch/conf
 1. NO
 2. YES
[1-2, ENTER='NO']:NO

Error: Invalid option, input an integer
[1-2, ENTER='NO']:1

---------------------------------------------
Configuring T1/E1 cards [A101/A102/A104/A108]
---------------------------------------------
A101 detected on slot:4 bus:3

-----------------------------------------------------------
Configuring port 1 on A101 slot:4 bus:3.
-----------------------------------------------------------

Select media type for AFT-A101 on port 1 [slot:4 bus:3 span:1]
 1. T1
 2. E1
 3. Unused
 4. Exit
[1-4]:1

Configuring port 1 on AFT-A101 as: T1, coding:B8ZS, framing:ESF.
 1. YES - Keep these settings
 2. NO  - Configure line coding and framing
[1-2, ENTER='YES']:YES

Error: Invalid option, input an integer
[1-2, ENTER='YES']:1

Select clock for AFT-A101 on port 1 [slot:4 bus:3 span:1]
 1. NORMAL
 2. MASTER
[1-2]:1

Select Switchtype for AFT-A101 on port 1 [slot:4 bus:3 span:1]
 1. National
 2. Nortel DMS100
 3. Lucent 5ESS
 4. Lucent 4ESS
[1-4]:1

Select signalling type for AFT-A101 on port 1 [slot:4 bus:3 span:1]
 1. PRI CPE
 2. PRI NET
[1-2]:1

Select dialplan context for AFT-A101 on port 1
 1. default
 2. public
 3. Custom
[1-3]:1


Input the dialing group for this port
: 1

Would you like to enable hardware DTMF detection?
 1. YES
 2. NO
[1-2, ENTER='YES']:1

Would you like to enable hardware fax detection?
 1. YES
 2. NO
[1-2, ENTER='NO']:1


Port 1 on AFT-A101 configuration complete...
Press any key to continue:

T1/E1 card configuration complete.
Press any key to continue:

------------------------------------
Configuring ISDN BRI cards [A500/B700]
------------------------------------

No Sangoma ISDN BRI cards detected

Press any key to continue:
------------------------------------
Configuring GSM cards [W400]
------------------------------------

No Sangoma GSM cards detected

Press any key to continue:

------------------------------------
Configuring analog cards [A200/A400/B600/B610/B700/B800]
------------------------------------
------------------------------------
Configuring USB devices [U100]
------------------------------------

###################################################################
#                             SUMMARY                             #
###################################################################

  1 T1/E1 port(s) detected, 1 configured
  0 ISDN BRI port(s) detected, 0 configured
  0 analog card(s) detected, 0 configured
  0 GSM card(s) detected, 0 configured
  0 usb device(s) detected, 0 configured

Configurator will create the following files:
    1. Wanpipe config files in /etc/wanpipe
    2. freetdm config file /usr/local/freeswitch/conf/freetdm.conf
    3. freetdm_xml config file /usr/local/freeswitch/conf/freetdm.conf.xml

Your configuration has been saved in /etc/wanpipe/debug-2012-05-18.tgz.
When requesting support, email this file to techdesk at sangoma.com<mailto:techdesk at sangoma.com>


###################################################################

Configuration Complete! Please select following:
 1. YES - Continue
 2. NO - Exit
[1-2]:1



 Wanpipe configuration complete: choose action
 1. Save cfg: Stop Wanpipe now
 2. Do not save cfg: Exit
[1-2]:1

sh: asterisk: not found
sh: asterisk: not found

Stopping Wanpipe...


Removing old configuration files...

Copying new Wanpipe configuration files...

Copying new freetdm configuration files (/usr/local/freeswitch/conf/freetdm.conf)...

Copying new freetdm configuration files (/usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml)...

Wanrouter start complete...
Current boot level is 2

Wanrouter boot scripts configuration...

Removing existing wanrouter boot scripts...OK
Would you like wanrouter to start on system boot?
 1. YES
 2. NO
[1-2]:1

Verifying Zaptel boot scripts...
Enabling wanrouter init scripts ...(start:8, stop:91)
Sangoma cards configuration complete, exiting...

root at copycall:~#

thanks,
dave cook


On Mon, May 28, 2012 at 12:24 PM, Moises Silva <moises.silva at gmail.com<mailto:moises.silva at gmail.com>> wrote:
On Mon, May 28, 2012 at 12:49 AM, copycall <dave at copycall.com<mailto:dave at copycall.com>> wrote:
curriegrad2004,

thanks for responding.

unfortunately, the sangoma tech support has not been as crack as usual.

i can go into the specifics, but it probably isn't necessary in a public forum. i'm not looking to piss anyone off.

if you have you actually configured a a101 sangoma card for a legacy CAS T1 card, i would greatly appreciate your assistance.


It's been a while since I configured for RBS, but I may be able to give you some pointers. Where are you at? did you have the drivers compiled?

http://wiki.sangoma.com/wanpipe-freeswitch-ftdm

As that web page states, RBS with FreeSWITCH is only supported using Wanpipe + DAHDI, not in native Wanpipe mode. http://wiki.sangoma.com/FreeSWITCH-Dahdi-Mode

The module you want to use is ftmod_analog



Moises Silva
Manager, Software Engineering

msilva at sangoma.com<mailto:msilva at sangoma.com>

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475<tel:%2B1%20800%20388%202475> (N. America)

t.   +1 905 474 1990 x128<tel:%2B1%20905%20474%201990%20x128>

f.   +1 905 474 9223<tel:%2B1%20905%20474%209223>




<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>


Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>


 <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

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




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

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
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From acarrega at vartel.com  Fri Sep  7 23:26:47 2012
From: acarrega at vartel.com (Andrew Carrega)
Date: Fri, 7 Sep 2012 19:26:47 +0000
Subject: [Freeswitch-users] Question about Exporting the cacaert.pem for
	client devices
Message-ID: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039B9D1@SN2PRD0610MB384.namprd06.prod.outlook.com>

I followed the Freeswitch wiki for enabling tls & srtp on Freeswitch. I have it enabled on my internal and external profiles and both profiles are starting up just fine.
I can review my certificate details with the command:
openssl x509 -noout -inform pem -text -in /usr/local/freeswitch/conf/ssl/agent.pem

I am not at this section of the wiki where it says the clients should have at least the CA root certificate.
Clients should all have at least the CA root certificate installed onto them in order to ensure security. Without enabling chain verification (that the server certificate was issued by the approved CA) a MITM attack is possible against a client. The CA certificate is the conf/ssl/cafile.pem it contains only a certificate and clients use it to ensure the server certificate is issued by the CA.

Where I am stuck is understanding how to export or download the cacert.pem from the server? I seem to don't understand the process or tools to use and I can't seem to access /usr/local/freeswitch/conf/ssl  directory or the /usr/local/freeswitch/conf/ssl/CA from root.

Any help is appreciated.




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From msc at freeswitch.org  Fri Sep  7 23:37:33 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Sep 2012 12:37:33 -0700
Subject: [Freeswitch-users] mod_perl installation error
In-Reply-To: <1347032281.54902.YahooMailNeo@web111611.mail.gq1.yahoo.com>
References: <1347032281.54902.YahooMailNeo@web111611.mail.gq1.yahoo.com>
Message-ID: <CAKzWOxV1QJtH3LNv46O_99XuWzszyy-hugN5jiiurd0xyNQjhA@mail.gmail.com>

Please open a ticket at jira.freeswitch.org and include this information.
Thanks!
-MC

On Fri, Sep 7, 2012 at 8:38 AM, Hardik Patel <hard_patel09 at yahoo.com> wrote:

> Hi,
>
> I am doing installation of freeswitch with mod_perl module but during
> compilation i got below error.
>
> ********ERROR**************
> making all mod_perl
> Creating mod_perl.la...
> /usr/bin/ld:
> /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a(op.o): relocation
> R_X86_64_32S against `PL_sv_yes' can not be used when making a shared
> object; recompile with -fPIC
> /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a: could not read
> symbols: Bad value
> collect2: ld returned 1 exit status
> cat: .libs/mod_perl.log: No such file or directory
> make[5]: *** [mod_perl.la] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_perl-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> ***********************************************
>
> making all mod_perl
> Creating mod_perl.la...
> /usr/bin/ld:
> /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a(op.o): relocation
> R_X86_64_32S against `PL_sv_yes' can not be used when making a shared
> object; recompile with -fPIC
> /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a: could not read
> symbols: Bad value
> collect2: ld returned 1 exit status
> cat: .libs/mod_perl.log: No such file or directory
> make[5]: *** [mod_perl.la] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_perl-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
> can anybody please help me to figure out the issue?
>
> Thanks in advance.
>
> Thanks,
> Hardik Patel
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep  7 23:45:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Sep 2012 12:45:18 -0700
Subject: [Freeswitch-users] More information on FreeSwitch
In-Reply-To: <1347034731293-7582678.post@n2.nabble.com>
References: <CC20783F-329B-4377-914E-CE4D640529C6@gmail.com>
	<CACc3D0XO5kOZ_imMy6XazzTGyYUpgJtrRRoAtkK9_WKTbFFo+w@mail.gmail.com>
	<CALNZuwnv1SzQ8FNC47U7SZuwAFo7u6JK3mxaGTmmOkVOfMF=rQ@mail.gmail.com>
	<1347034731293-7582678.post@n2.nabble.com>
Message-ID: <CAKzWOxX3gV9GABaqWbZ6_=jRPOWXxk-fDM+KMnGRe3wx7b890w@mail.gmail.com>

>
> I bought the original FreeSwitch book in addition to the cookbook and while
> it is an excellent text with a lot of good information contained therein,
> there's still some info that's not in there that I'd like to see.  For
> instance, I'm still left with questions about FS' capabilities and behavior
> with NAT handling in certain scenarios and that info doesn't appear to be
> very well covered anywhere that I've been able to find - probably due to
> the
> fact that FS is still a work in progress.  Not many people seem to truly
> understand the details of FS' NAT functionality, so unless you're lucky and
> one of the developers who truly understands it has the time to answer your
> question, you'll have to wade through the code and experiment (which isn't
> always a bad thing, another one of the many advantages of open source) but
> that can be quite time-consuming.
>
This is one reason we are doing a second edition to the "bridge book." We
are planning on having an entire chapter dedicated just to handling NAT.

>
> I was surprised to find that the full original FS book is floating around
> the 'Net in PDF form, not sure that the authors or the publisher would be
> particularly happy about that.  If any of the AUTHORS want the link(s),
> contact me and I'll give the info I have to you.  If your name doesn't
> appear on the cover of the FS book however, don't ask me for any links.
> :)
>
> As one of the co-authors I can honestly say that I'm not the least
concerned about copies of the book floating around. (Disclaimer: I speak
only for my self, not for the publisher or my fellow authors.) True fans of
the FreeSWITCH project will purchase the book regardless of whether or not
it is available for "free" online. Time and energy spent "fighting piracy"
is time and energy not spent making the 2e book even better. </soapbox>

Thanks for the info and feedback!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From paul at cupis.co.uk  Fri Sep  7 23:47:53 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Fri, 07 Sep 2012 20:47:53 +0100
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
Message-ID: <504A4F69.60901@cupis.co.uk>

On 06/09/12 20:09, Matt Putnam wrote:
> Sorry for the confusion I can really only equate it to asterisk as
> that?s what our current platform is. Essentially the problem is that if
> I have a PBX that has a trunk that is registered to freeswitch and a
> call is received for a DID that is on that trunk I am currently unable
> to send that DID to the PBX. When the call is received on the PBX side
> it looks as if the call is for the trunk instead of the DID is should be
> for. So in the SIP invite instead of using lightspar1 at domain it would
> use 5551212 at domain. In asterisk this was simply accomplished by a dial
> statement of (SIP/5551212 at lightspar1) what I am looking for is that
> equivalent in freeswitch.

Do you have a copy of the FreeSWITCH Cookbook? This requirement sounds 
similar to one of the recipes in the book.

<extension name="9725551212">
   <condition field="destination_number" expression="^9725551212$"/>

   <condition field="${sofia_contact(lightspar1@${domain})}"
                expression="^[^\@]+(.*)">
     <action application="bridge" data="sofia/external/9725551212$1"/>
   </condition>
</extension>

The sofia_contact bit should work out the dialstring and then strip 
everything before the '@' (in your example this would be 'lightspar1') 
and then the bridge prepends the destination number.

Regards,


From mitch.capper at gmail.com  Sat Sep  8 06:08:40 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 7 Sep 2012 19:08:40 -0700
Subject: [Freeswitch-users] Question about Exporting the cacaert.pem for
 client devices
In-Reply-To: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039B9D1@SN2PRD0610MB384.namprd06.prod.outlook.com>
References: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039B9D1@SN2PRD0610MB384.namprd06.prod.outlook.com>
Message-ID: <CALn233pGtkcGVQ+M2EF1iGV5=rCW-fjASdxyFBGx90BtnKERfw@mail.gmail.com>

Sure so the cafile.pem should only contain a "BEGIN CERTIFICATE" and
"END CERTIFICATE" block no PRIVATE KEY.   You can copy this file and
most clients will expect a .crt file, you can just rename it from
cafile.pem to ca.crt.   As for loading it into a specific client that
will depend on the sip client.  If its a softphone it may trust any CA
installed in the windows certificate store, in which case you can
double click and open the .crt file in windows and just import it.
Otherwise search for the phone and "server certificate" or "ca
certificate" and import and it should have details.


~Mitch

On Fri, Sep 7, 2012 at 12:26 PM, Andrew Carrega <acarrega at vartel.com> wrote:
> I followed the Freeswitch wiki for enabling tls & srtp on Freeswitch. I have
> it enabled on my internal and external profiles and both profiles are
> starting up just fine.
>
> I can review my certificate details with the command:
>
> openssl x509 -noout -inform pem -text -in
> /usr/local/freeswitch/conf/ssl/agent.pem
>
>
>
> I am not at this section of the wiki where it says the clients should have
> at least the CA root certificate.
>
> Clients should all have at least the CA root certificate installed onto them
> in order to ensure security. Without enabling chain verification (that the
> server certificate was issued by the approved CA) a MITM attack is possible
> against a client. The CA certificate is the conf/ssl/cafile.pem it contains
> only a certificate and clients use it to ensure the server certificate is
> issued by the CA.
>
>
>
> Where I am stuck is understanding how to export or download the cacert.pem
> from the server? I seem to don?t understand the process or tools to use and
> I can?t seem to access /usr/local/freeswitch/conf/ssl  directory or the
> /usr/local/freeswitch/conf/ssl/CA from root.
>
>
>
> Any help is appreciated.
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep  8 06:33:51 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 7 Sep 2012 22:33:51 -0400
Subject: [Freeswitch-users] Registering SIP Trunks to Freeswitch
In-Reply-To: <504A4F69.60901@cupis.co.uk>
References: <9757304EEA8CE74494B6D21E63DA005212AE44CFC5@MBX23.exg5.exghost.com>
	<CAKzWOxXs6zktidb6LcFHv6Cwtb=DnzNthD0vENbBQL4wXjy4pA@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D0A2@MBX23.exg5.exghost.com>
	<CAKV-jFyt5=sPNkhTfNBV7xfZCmXzBOV++L=87MNet37hk6U70w@mail.gmail.com>
	<9757304EEA8CE74494B6D21E63DA005212AE44D1B0@MBX23.exg5.exghost.com>
	<504A4F69.60901@cupis.co.uk>
Message-ID: <CAKV-jFzxP-JTf2dsRVOgY94imNQbVfAAV6uq21jQcz40gKc=AA@mail.gmail.com>

If the other PBX is FreeSWITCH you need.to rewrite the Request URI before
sending to the receiving FS box. Sending the username before the @ in the
request URI is done by default but if you need to differentiate DIDs then
you have to rewrite the request URI to that DID so it does show up in
DESTINATION_NUMBER. Search RURI on the wiki. If it's coming from another
PBX that's not FS we can't help and if that's coming from your provider its
time for another provider.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 7, 2012 3:48 PM, "Paul Cupis" <paul at cupis.co.uk> wrote:

> On 06/09/12 20:09, Matt Putnam wrote:
> > Sorry for the confusion I can really only equate it to asterisk as
> > that?s what our current platform is. Essentially the problem is that if
> > I have a PBX that has a trunk that is registered to freeswitch and a
> > call is received for a DID that is on that trunk I am currently unable
> > to send that DID to the PBX. When the call is received on the PBX side
> > it looks as if the call is for the trunk instead of the DID is should be
> > for. So in the SIP invite instead of using lightspar1 at domain it would
> > use 5551212 at domain. In asterisk this was simply accomplished by a dial
> > statement of (SIP/5551212 at lightspar1) what I am looking for is that
> > equivalent in freeswitch.
>
> Do you have a copy of the FreeSWITCH Cookbook? This requirement sounds
> similar to one of the recipes in the book.
>
> <extension name="9725551212">
>    <condition field="destination_number" expression="^9725551212$"/>
>
>    <condition field="${sofia_contact(lightspar1@${domain})}"
>                 expression="^[^\@]+(.*)">
>      <action application="bridge" data="sofia/external/9725551212$1"/>
>    </condition>
> </extension>
>
> The sofia_contact bit should work out the dialstring and then strip
> everything before the '@' (in your example this would be 'lightspar1')
> and then the bridge prepends the destination number.
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep  8 07:09:43 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 7 Sep 2012 23:09:43 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <504A072C.2000101@puzzled.xs4all.nl>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
	<1347016045.19904.64.camel@luna.madrid.commsmundi.com>
	<5049E4B2.5090205@puzzled.xs4all.nl>
	<1347024299.19904.72.camel@luna.madrid.commsmundi.com>
	<504A072C.2000101@puzzled.xs4all.nl>
Message-ID: <CAKV-jFyLXAJLPg2b3e0MPsfqC_vm=qqb1a93zDVXo4Xn05-pkw@mail.gmail.com>

I use Polycoms with all new deployments. Linksys/Cisco SPA stuff is nice
too, but we stopped using them. One reason was because of Polycom's UC
Firmware and the possibilities when using it. Besides that, they have a
more 'polished' feel to them and the speakerphones are unbelievable, even
with their entry class phones.

Cisco 79XX series are absolute crap. They can't be trusted, not when using
SIP. Grandstream makes OK phones, probably not ready for primetime
(although I dig their ATAs because they just work). Aastras have indeed
some bugs that are annoying. Haven't really tried Snom well enough to give
a good or bad review.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Sep 7, 2012 10:41 AM, "Patrick Lists" <freeswitch-list at puzzled.xs4all.nl>
wrote:

> On 09/07/2012 03:24 PM, Fran?ois Delawarde wrote:
> > Hi Patrick,
> >
> > Not sure I understand the question. Aren't all LCD screens backlit?
>
> The Cisco 7961 in my home office does not have a backlit LCD screen
> (like the Polycom IP670). Quite annoying in low light conditions so it's
> gathering dust.
>
> > But yes the screen is a 320x160 LCD with backlight. About keys, you have
> > 6 programmable keys on the top part and 10 more on the right with LEDs
> > for BLF or other things. The other keys (dtmf, navigation, volume, etc.)
> > have no light.
>
> Thanks for the 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 asaad2 at gmail.com  Sat Sep  8 15:39:52 2012
From: asaad2 at gmail.com (BookBag)
Date: Sat, 8 Sep 2012 07:39:52 -0400
Subject: [Freeswitch-users] Best SIP phone?
In-Reply-To: <CAKV-jFyLXAJLPg2b3e0MPsfqC_vm=qqb1a93zDVXo4Xn05-pkw@mail.gmail.com>
References: <27583F77-C819-4CC7-96A0-A1680435B18A@kavun.ch>
	<504935A0.9070106@communicatefreely.net>
	<8D3EF6E0-274B-44F5-B6F9-9E40276C89FD@kavun.ch>
	<5049423B.9070401@communicatefreely.net>
	<1347007356.19904.43.camel@luna.madrid.commsmundi.com>
	<F9DCABD0-A6F3-463A-B6C7-80B4E56BF4F3@kavun.ch>
	<1347016045.19904.64.camel@luna.madrid.commsmundi.com>
	<5049E4B2.5090205@puzzled.xs4all.nl>
	<1347024299.19904.72.camel@luna.madrid.commsmundi.com>
	<504A072C.2000101@puzzled.xs4all.nl>
	<CAKV-jFyLXAJLPg2b3e0MPsfqC_vm=qqb1a93zDVXo4Xn05-pkw@mail.gmail.com>
Message-ID: <CAAdTMa_iWSNFyZscJ7DgKNF3k8n=khkwrbS1cXpTExqHzjXk2g@mail.gmail.com>

I can vouch for cisco under sccp but under sip its sub par. And don't
forget that you need smartnet in order to download the sip firmware which
is always lacking in features than the current sip protocol. By the time
they finally implement the complete feature set you'll find the sip
protocol advanced and now has more features. But you can't download their
new sip firmware anyway because you need to consistently pay for smartnet
so you can download it.

On Sep 7, 2012 11:11 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
> I use Polycoms with all new deployments. Linksys/Cisco SPA stuff is nice
too, but we stopped using them. One reason was because of Polycom's UC
Firmware and the possibilities when using it. Besides that, they have a
more 'polished' feel to them and the speakerphones are unbelievable, even
with their entry class phones.
>
> Cisco 79XX series are absolute crap. They can't be trusted, not when
using SIP. Grandstream makes OK phones, probably not ready for primetime
(although I dig their ATAs because they just work). Aastras have indeed
some bugs that are annoying. Haven't really tried Snom well enough to give
a good or bad review.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Sep 7, 2012 10:41 AM, "Patrick Lists" <
freeswitch-list at puzzled.xs4all.nl> wrote:
>>
>> On 09/07/2012 03:24 PM, Fran?ois Delawarde wrote:
>> > Hi Patrick,
>> >
>> > Not sure I understand the question. Aren't all LCD screens backlit?
>>
>> The Cisco 7961 in my home office does not have a backlit LCD screen
>> (like the Polycom IP670). Quite annoying in low light conditions so it's
>> gathering dust.
>>
>> > But yes the screen is a 320x160 LCD with backlight. About keys, you
have
>> > 6 programmable keys on the top part and 10 more on the right with LEDs
>> > for BLF or other things. The other keys (dtmf, navigation, volume,
etc.)
>> > have no light.
>>
>> Thanks for the 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 mkdutchman at gmail.com  Sat Sep  8 14:17:57 2012
From: mkdutchman at gmail.com (Melvin King)
Date: Sat, 8 Sep 2012 06:17:57 -0400
Subject: [Freeswitch-users] Odd SIP user registration problem
Message-ID: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>

Hello,
I've been beating my brains out for a while with this problem, I think I
might need some help. I've been trying to set it up as a trimmed down to
the max carrier grade SIP PBX.

Here's my setup

I'm using a amazon ec2 64 bit image with a public static IP for the host.
(Inside a NAT)

ports opened from the outside are
TCP 5060
UDP 16384 - 32768

FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2;
git at commit e97da8e20a

This is the relevant error message that I am getting, I cannot register any
users at all, even though they're in the directory

2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [
1000 at 23.21.191.100]
2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [
1000 at 23.21.191.100]
You must define a domain called '23.21.191.100' in your directory and add a
user with the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [
1000 at 23.21.191.100]
2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [
1000 at 23.21.191.100]
2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded flowroute
with code 200 - count -1/1/1, state UP
2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [
1000 at 23.21.191.100]
You must define a domain called '23.21.191.100' in your directory and add a
user with the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [
1000 at 23.21.191.100]


Any insight would be greatly appreciated

Mel
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From chris at gonumina.com  Sat Sep  8 19:55:10 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Sat, 8 Sep 2012 11:55:10 -0400
Subject: [Freeswitch-users] Odd SIP user registration problem
In-Reply-To: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>
References: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>
Message-ID: <8333860497797977792@unknownmsgid>

See what that IP is 23.21.191.100.


It looks like this is a STUN Server issue on the Sip Phones.

___________________
Mobile Reply

On Sep 8, 2012, at 9:16 AM, Melvin King <mkdutchman at gmail.com> wrote:

Hello,
I've been beating my brains out for a while with this problem, I think I
might need some help. I've been trying to set it up as a trimmed down to
the max carrier grade SIP PBX.

Here's my setup

I'm using a amazon ec2 64 bit image with a public static IP for the host.
(Inside a NAT)

ports opened from the outside are
TCP 5060
UDP 16384 - 32768

FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2;
git at commit e97da8e20a

This is the relevant error message that I am getting, I cannot register any
users at all, even though they're in the directory

2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [
1000 at 23.21.191.100]
2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [
1000 at 23.21.191.100]
You must define a domain called '23.21.191.100' in your directory and add a
user with the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [
1000 at 23.21.191.100]
2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [
1000 at 23.21.191.100]
2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded flowroute
with code 200 - count -1/1/1, state UP
2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [
1000 at 23.21.191.100]
You must define a domain called '23.21.191.100' in your directory and add a
user with the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [
1000 at 23.21.191.100]


Any insight would be greatly appreciated

Mel

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From acarrega at vartel.com  Sat Sep  8 20:01:53 2012
From: acarrega at vartel.com (Andrew Carrega)
Date: Sat, 8 Sep 2012 16:01:53 +0000
Subject: [Freeswitch-users] Question about Exporting the cacaert.pem
	for	client devices
In-Reply-To: <CALn233pGtkcGVQ+M2EF1iGV5=rCW-fjASdxyFBGx90BtnKERfw@mail.gmail.com>
References: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039B9D1@SN2PRD0610MB384.namprd06.prod.outlook.com>,
	<CALn233pGtkcGVQ+M2EF1iGV5=rCW-fjASdxyFBGx90BtnKERfw@mail.gmail.com>
Message-ID: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039BFD9@SN2PRD0610MB384.namprd06.prod.outlook.com>

Hi Mitch,
thanks for your help. You say below copy this ca.cert.pem file. "that's where I am stuck" are you suggesting I can copy and paste the contents from this file or "copy the file" off of the freeswitch server. If you are saying copy the ca.cert.pem file off the server that is where I am stuck. I can't get to that directory /usr/local/freeswitch/conf/ssl/ to copy the file.  I can view the contents of the ca.cert.pem file with the 

openssl x509 -noout -inform pem -text -in /usr/local/freeswitch/conf/ssl/CA/ca.cert.pem comand but that is all I am able to do at the moment.

________________________________________
From: Mitch Capper [mitch.capper at gmail.com]
Sent: Friday, September 07, 2012 10:08 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Question about Exporting the cacaert.pem for    client devices

Sure so the cafile.pem should only contain a "BEGIN CERTIFICATE" and
"END CERTIFICATE" block no PRIVATE KEY.   You can copy this file and
most clients will expect a .crt file, you can just rename it from
cafile.pem to ca.crt.   As for loading it into a specific client that
will depend on the sip client.  If its a softphone it may trust any CA
installed in the windows certificate store, in which case you can
double click and open the .crt file in windows and just import it.
Otherwise search for the phone and "server certificate" or "ca
certificate" and import and it should have details.


~Mitch

On Fri, Sep 7, 2012 at 12:26 PM, Andrew Carrega <acarrega at vartel.com> wrote:
> I followed the Freeswitch wiki for enabling tls & srtp on Freeswitch. I have
> it enabled on my internal and external profiles and both profiles are
> starting up just fine.
>
> I can review my certificate details with the command:
>
> openssl x509 -noout -inform pem -text -in
> /usr/local/freeswitch/conf/ssl/agent.pem
>
>
>
> I am not at this section of the wiki where it says the clients should have
> at least the CA root certificate.
>
> Clients should all have at least the CA root certificate installed onto them
> in order to ensure security. Without enabling chain verification (that the
> server certificate was issued by the approved CA) a MITM attack is possible
> against a client. The CA certificate is the conf/ssl/cafile.pem it contains
> only a certificate and clients use it to ensure the server certificate is
> issued by the CA.
>
>
>
> Where I am stuck is understanding how to export or download the cacert.pem
> from the server? I seem to don?t understand the process or tools to use and
> I can?t seem to access /usr/local/freeswitch/conf/ssl  directory or the
> /usr/local/freeswitch/conf/ssl/CA from root.
>
>
>
> Any help is appreciated.
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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.burlingame at me.com  Sat Sep  8 20:36:00 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sat, 08 Sep 2012 09:36:00 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
Message-ID: <C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>

Ok got another crash oddly enough it's kinda the same as yesterday's 

Yesterday the last message I see in the log file is a 401 Unauthorized before the log file stopped recording data and fs_cli exited to prompt today same thing happened but this time it looks like it did a core dump.

Opening JIRA case now


On Sep 7, 2012, at 10:23 AM, Mike Burlingame <mike.burlingame at me.com> wrote:

> I figured I would update to the latest git due to in the past I have seen "make sure your running the latest git" the odd thing is no core dump was created on the last crash.
> 
> I will post the backtrace if a core dump is generated after the next crash
> 
> 
> On Sep 7, 2012, at 10:16 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> FYI: You need to get the backtrace before you update or the core file becomes useless.
>> It dumps core by default so you should have one but if you updated, its tool late unless you can checkout the exact previous rev.
>> 
>> 
>> 
>> On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> after putting the server in production and  loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.
>> 
>> However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.
>> 
>> 
>> 
>> On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> 
>> > After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
>> >
>> > Thanks
>> >
>> > On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> >
>> >> Looks much much better Thank you -- Now to conduct more testing
>> >>
>> >> 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-Leg:5060, with session description
>> >> 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying
>> >> 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>> >> 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>> >> 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >> 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
>> >> 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >> 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >> 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description
>> >> 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
>> >> 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> >> 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
>> >> 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK
>> >> 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> >> 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
>> >>
>> >>
>> >>
>> >> On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> >>
>> >>> ok,
>> >>>
>> >>> update one more time, if it still does not work just go right to jira
>> >>> with the latest (not before today's changes)
>> >>>
>> >>>
>> >>> On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> >>>> The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
>> >>>>
>> >>>> commit d45db898339e1b2212f5befff1af714abcec034f
>> >>>> Author: Anthony Minessale <anthm at freeswitch.org>
>> >>>> Date:   Wed Sep 5 13:11:32 2012 -0500
>> >>>>
>> >>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>> >>>> 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
>> >>>> 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>> >>>> 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>> >>>> 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>> 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>> >>>> 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>> 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
>> >>>> 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>> >>>> 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> >>>> 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> >>>> 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>> >>>> 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>> >>>> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
>> >>>> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>> >>>> 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>> >>>> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>> >>>> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>> >>>> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
>> >>>> 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
>> >>>> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
>> >>>> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>> >>>> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> >>>> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> >>>> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>> >>>>
>> >>>>
>> >>>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> >>>>
>> >>>>> update and try again, if it still doesn't work open a jira
>> >>>>>
>> >>>>>
>> >>>>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> >>>>>> as promised here is the update testing and enabling
>> >>>>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
>> >>>>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
>> >>>>>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>> >>>>>>
>> >>>>>> I can open a JIRA case on this and provide the console log file / PCAP's ect
>> >>>>>> if that would help
>> >>>>>>
>> >>>>>>
>> >>>>>> Call Flow with out {sip_wait_for_aleg_ack=true}
>> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
>> >>>>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>> >>>>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>> >>>>>> sip:13605551212 at B-Leg, with session description
>> >>>>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>> >>>>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>> >>>>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> >>>>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> >>>>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> >>>>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>> >>>>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> >>>>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>> >>>>>>
>> >>>>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
>> >>>>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>> >>>>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>> >>>>>> sip:13605551212 at B-Leg, with session description
>> >>>>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>> >>>>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>> >>>>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>> >>>>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>> >>>>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>> >>>>>> description
>> >>>>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> >>>>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>> >>>>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>> >>>>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>> >>>>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>> >>>>>>
>> >>>>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> >>>>>>
>> >>>>>> No worries I will be out this weekend for the long weekend I will work on
>> >>>>>> getting the test box upgraded and a test case setup on Tuesday I will report
>> >>>>>> back the results mid to late next week and provided everything works as I
>> >>>>>> hope it will I will be happy to pay the Wiki tax :)
>> >>>>>>
>> >>>>>>
>> >>>>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
>> >>>>>> wrote:
>> >>>>>>>
>> >>>>>>> Cool I will nail that up on my test box and see if that works
>> >>>>>>>
>> >>>>>> Please report back on whether it works or not and then be prepared to pay
>> >>>>>> the wiki tax. :) I'll be glad to assist with getting this documented
>> >>>>>> although I think you're in the best position to give that documentation some
>> >>>>>> real-world context.
>> >>>>>>
>> >>>>>> -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
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> _________________________________________________________________________
>> >>>>>> Professional FreeSWITCH Consulting Services:
>> >>>>>> consulting at freeswitch.org
>> >>>>>> http://www.freeswitchsolutions.com
>> >>>>>>
>> >>>>>> 
>> >>>>>> 
>> >>>>>>
>> >>>>>> Official FreeSWITCH Sites
>> >>>>>> http://www.freeswitch.org
>> >>>>>> http://wiki.freeswitch.org
>> >>>>>> http://www.cluecon.com
>> >>>>>>
>> >>>>>> FreeSWITCH-users mailing list
>> >>>>>> FreeSWITCH-users at 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
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 openser at yeah.net  Sat Sep  8 20:01:16 2012
From: openser at yeah.net (openser)
Date: Sun, 9 Sep 2012 00:01:16 +0800 (CST)
Subject: [Freeswitch-users] record video to file  of a two way call
Message-ID: <76af848f.596f.139a69d1745.Coremail.openser@yeah.net>

Hi,
I want to record video to a file of a bridged video call, i kown record_session app ,but it only record audio, and record_fsv app only record one call leg but not a two leg call, is there any app or what can i do for this ? thanks.
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From acarrega at vartel.com  Sat Sep  8 23:41:35 2012
From: acarrega at vartel.com (Andrew Carrega)
Date: Sat, 8 Sep 2012 19:41:35 +0000
Subject: [Freeswitch-users] Odd SIP user registration problem
In-Reply-To: <8333860497797977792@unknownmsgid>
References: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>,
	<8333860497797977792@unknownmsgid>
Message-ID: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039C108@SN2PRD0610MB384.namprd06.prod.outlook.com>

What do you have set for your external_rtp_ip and external_sip_ip??
________________________________
From: Chris Ferreira [chris at gonumina.com]
Sent: Saturday, September 08, 2012 11:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Odd SIP user registration problem

See what that IP is 23.21.191.100.


It looks like this is a STUN Server issue on the Sip Phones.

___________________
Mobile Reply

On Sep 8, 2012, at 9:16 AM, Melvin King <mkdutchman at gmail.com<mailto:mkdutchman at gmail.com>> wrote:

Hello,
I've been beating my brains out for a while with this problem, I think I might need some help. I've been trying to set it up as a trimmed down to the max carrier grade SIP PBX.

Here's my setup

I'm using a amazon ec2 64 bit image with a public static IP for the host. (Inside a NAT)

ports opened from the outside are
TCP 5060
UDP 16384 - 32768

FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2; git at commit e97da8e20a

This is the relevant error message that I am getting, I cannot register any users at all, even though they're in the directory

2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [1000 at 23.21.191.100<mailto:1000 at 23.21.191.100>]
2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [1000 at 23.21.191.100<mailto:1000 at 23.21.191.100>]
You must define a domain called '23.21.191.100' in your directory and add a user with the id="1000" attribute
and you must configure your device to use the proper domain in it's authentication credentials.
2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [1000 at 23.21.191.100<mailto:1000 at 23.21.191.100>]
2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [1000 at 23.21.191.100<mailto:1000 at 23.21.191.100>]
2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded flowroute with code 200 - count -1/1/1, state UP
2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [1000 at 23.21.191.100<mailto:1000 at 23.21.191.100>]
You must define a domain called '23.21.191.100' in your directory and add a user with the id="1000" attribute
and you must configure your device to use the proper domain in it's authentication credentials.
2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [1000 at 23.21.191.100<mailto:1000 at 23.21.191.100>]


Any insight would be greatly appreciated

Mel
_________________________________________________________________________
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 anthony.minessale at gmail.com  Sun Sep  9 01:27:32 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 8 Sep 2012 16:27:32 -0500
Subject: [Freeswitch-users] Confusion about sip hangup cause Q850 hangup
 cause and long struggles with a provider.
In-Reply-To: <22952124.18075.1347007177101.JavaMail.root@mail>
References: <1407731.17913.1347006256794.JavaMail.root@mail>
	<22952124.18075.1347007177101.JavaMail.root@mail>
Message-ID: <CAKbxfG-w6N2RvQ5SvAgh0qnCyaSRq=BDnL7MWaERevM4P86k-g@mail.gmail.com>

I think our wiki is less accurate than our code:  Our cause mappings are
straight from RFC 4497 http://tools.ietf.org/html/rfc4497

Here is an excerpt:

 408 Request timeout                 102 Recovery on timer expiry

 504 Gateway time-out                102 Recovery on timer expiry





On Fri, Sep 7, 2012 at 3:39 AM, Sias Mey <sias at cpdata.co.za> wrote:

> Hi, Im sure this subject has been beaten to death .. but many googles and
> many email searches hasent really managed to find me something.
>
> Im a dev for a small company that writes call center software. Freeswitch
> was a godsend, thank you.
>
> Now .. the confusion.
>
> We are getting a lot of what seem to be strange hangup codes from a new
> provider big fights about loads of failled calls ensued blah blah.. much
> sip packet logging and manual inspection later.. I found the following.
>
> from xmlcdr.
>
>
>     <sip_hangup_disposition>recv_refuse</sip_hangup_disposition>
>     <sip_term_status>408</sip_term_status>
>     <proto_specific_hangup_cause>sip%3A408</proto_specific_hangup_cause>
>     <sip_term_cause>102</sip_term_cause>
>     <hangup_cause>RECOVERY_ON_TIMER_EXPIRE</hangup_cause>
>     <hangup_cause_q850>102</hangup_cause_q850>
>
> this just to show its the same call
>     <sip_call_id>0d4d4e76-735a-1230-d2ac-000423b5571b</sip_call_id>
>
> and from the sip messages.
>
> SIP/2.0 408 Request Timeout
> Call-ID: 0d4d4e76-735a-1230-d2ac-000423b5571b
> Reason: Q.850;cause=18;text="no user responding"
>
> And according to the very useful wiki page on Q.850 codes 408 should = 18
> like it does in the providers response.
>
> Why then is the q850 hangup cause in the CDR 102? and where does that
> translation come from.
>
> This is a single example but I also have loads and loads where the CDR
> claims q850 code 18 but the sip messages provide 31 or a range of other
> codes.
> I can understand if the q850 code from the sip message is not being read
> by FS since FS has to be a bit more agnostic than that and in the pas I
> have almost exclusively worked with direct connections to TDM hardware so
> my knowledge and understanding of the sip messages is rather limited. But
> even in that case, shouldent the q850 code in the cdr at least conform to
> the translation from the wiki page?
>
> Oh I am not currently running the latest git release, having some libtiff
> issues on ubuntu to compile. I will respond to this again if I manage that
> and it helps matters.
>
> Thank you for your time and help,
> Regards
> Sias
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mkdutchman at gmail.com  Sun Sep  9 06:02:08 2012
From: mkdutchman at gmail.com (Melvin King)
Date: Sat, 8 Sep 2012 22:02:08 -0400
Subject: [Freeswitch-users] Odd SIP user registration problem
In-Reply-To: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039C108@SN2PRD0610MB384.namprd06.prod.outlook.com>
References: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>
	<8333860497797977792@unknownmsgid>
	<D582AE1C5F25FE409F6BC8BAB69D9F4F3039C108@SN2PRD0610MB384.namprd06.prod.outlook.com>
Message-ID: <CALg5agZU4xXa3_n71OFPi0EqQDMOfw+ykYm-hwyiNXFMicGObQ@mail.gmail.com>

23.21.191.100 is the static address of the WAN side of the network that the
freeswitch host is connected to.

STUN is turned off.

23.21.191.100 is set as the external_rtp_ip and external_sip_ip

Mel

On Sat, Sep 8, 2012 at 3:41 PM, Andrew Carrega <acarrega at vartel.com> wrote:

>  What do you have set for your external_rtp_ip and external_sip_ip??
> ------------------------------
> *From:* Chris Ferreira [chris at gonumina.com]
> *Sent:* Saturday, September 08, 2012 11:55 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Odd SIP user registration problem
>
>   See what that IP is 23.21.191.100.
>
>
>  It looks like this is a STUN Server issue on the Sip Phones.
>
> ___________________
> Mobile Reply
>
> On Sep 8, 2012, at 9:16 AM, Melvin King <mkdutchman at gmail.com> wrote:
>
>   Hello,
> I've been beating my brains out for a while with this problem, I think I
> might need some help. I've been trying to set it up as a trimmed down to
> the max carrier grade SIP PBX.
>
> Here's my setup
>
> I'm using a amazon ec2 64 bit image with a public static IP for the host.
> (Inside a NAT)
>
> ports opened from the outside are
> TCP 5060
> UDP 16384 - 32768
>
> FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2;
> git at commit e97da8e20a
>
> This is the relevant error message that I am getting, I cannot register
> any users at all, even though they're in the directory
>
> 2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [
> 1000 at 23.21.191.100]
> 2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [
> 1000 at 23.21.191.100]
> You must define a domain called '23.21.191.100' in your directory and add
> a user with the id="1000" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [
> 1000 at 23.21.191.100]
> 2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [
> 1000 at 23.21.191.100]
> 2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded flowroute
> with code 200 - count -1/1/1, state UP
> 2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [
> 1000 at 23.21.191.100]
> You must define a domain called '23.21.191.100' in your directory and add
> a user with the id="1000" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [
> 1000 at 23.21.191.100]
>
>
> Any insight would be greatly appreciated
>
> Mel
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From josh at foshee.info  Sun Sep  9 06:40:43 2012
From: josh at foshee.info (Joshua Foshee)
Date: Sat, 8 Sep 2012 21:40:43 -0500
Subject: [Freeswitch-users] Call recording speed issue
Message-ID: <CAA=kHzLbobvFF737zV8Nn+DYrSu9dFommDG=o5ONeoatpPkBQA@mail.gmail.com>

We have a issue with the call quality of the recorded call to be in a
faster speed then what the call takes place in. I have took a packet
capture on the server to replay the audio stream and it sounds just fine.
Anyone have the same problem with the speed of the audio recording being
faster then what takes place. I have tried saving wav and mp3 on the
server. I prefer mp3 as far as size of files. But it happens with each of
them. It looks like the call lasted for 19 sec but the audio recording the
server saved was 13 sec. This seems to only happen on incoming calls from
sip gateway and not outgoing calls from extensions on the box. I have
included three file links.

One is the file from the server that freeswitch saved while recording

https://fm.ols18.com/filemanager/public/7dc3c2d38a3fe69e75869bd6efcade27.php?lang=en<http://www.filesend.net/download.php?f=63c35673fc1ddd4e4f2d6a95228f65fc>

Two is the audio file extract from the packet capture

*
https://fm.ols18.com/filemanager/public/f0d81c8694d7f68df5f921f54429a02c.php?lang=en
*
*
*
Three is the packet capture

https://fm.ols18.com/filemanager/public/3d079f5501f071f2f38e6c8458da0718.php?lang=en<http://www.filesend.net/download.php?f=d61ada05b405fc3b3c6750fe355bca94>


Anyone seen this or have any ideas on how to diagnose further?

Thanks in advance
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From chris at gonumina.com  Sun Sep  9 09:18:39 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Sun, 9 Sep 2012 01:18:39 -0400
Subject: [Freeswitch-users] Odd SIP user registration problem
In-Reply-To: <CALg5agZU4xXa3_n71OFPi0EqQDMOfw+ykYm-hwyiNXFMicGObQ@mail.gmail.com>
References: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>
	<8333860497797977792@unknownmsgid>
	<D582AE1C5F25FE409F6BC8BAB69D9F4F3039C108@SN2PRD0610MB384.namprd06.prod.outlook.com>
	<CALg5agZU4xXa3_n71OFPi0EqQDMOfw+ykYm-hwyiNXFMicGObQ@mail.gmail.com>
Message-ID: <-338675922493849483@unknownmsgid>

Where are your phones connecting from? Are they using domains in their
authentication?



___________________
Mobile Reply

On Sep 8, 2012, at 10:12 PM, Melvin King <mkdutchman at gmail.com> wrote:

23.21.191.100 is the static address of the WAN side of the network that the
freeswitch host is connected to.

STUN is turned off.

23.21.191.100 is set as the external_rtp_ip and external_sip_ip

Mel

On Sat, Sep 8, 2012 at 3:41 PM, Andrew Carrega <acarrega at vartel.com> wrote:

>  What do you have set for your external_rtp_ip and external_sip_ip??
> ------------------------------
> *From:* Chris Ferreira [chris at gonumina.com]
> *Sent:* Saturday, September 08, 2012 11:55 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Odd SIP user registration problem
>
>   See what that IP is 23.21.191.100.
>
>
>  It looks like this is a STUN Server issue on the Sip Phones.
>
> ___________________
> Mobile Reply
>
> On Sep 8, 2012, at 9:16 AM, Melvin King <mkdutchman at gmail.com> wrote:
>
>   Hello,
> I've been beating my brains out for a while with this problem, I think I
> might need some help. I've been trying to set it up as a trimmed down to
> the max carrier grade SIP PBX.
>
> Here's my setup
>
> I'm using a amazon ec2 64 bit image with a public static IP for the host.
> (Inside a NAT)
>
> ports opened from the outside are
> TCP 5060
> UDP 16384 - 32768
>
> FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2;
> git at commit e97da8e20a
>
> This is the relevant error message that I am getting, I cannot register
> any users at all, even though they're in the directory
>
> 2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [
> 1000 at 23.21.191.100]
> 2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [
> 1000 at 23.21.191.100]
> You must define a domain called '23.21.191.100' in your directory and add
> a user with the id="1000" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [
> 1000 at 23.21.191.100]
> 2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [
> 1000 at 23.21.191.100]
> 2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded flowroute
> with code 200 - count -1/1/1, state UP
> 2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [
> 1000 at 23.21.191.100]
> You must define a domain called '23.21.191.100' in your directory and add
> a user with the id="1000" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [
> 1000 at 23.21.191.100]
>
>
> Any insight would be greatly appreciated
>
> Mel
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Sun Sep  9 09:54:51 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Sun, 09 Sep 2012 08:54:51 +0300
Subject: [Freeswitch-users] Call recording speed issue
In-Reply-To: <CAA=kHzLbobvFF737zV8Nn+DYrSu9dFommDG=o5ONeoatpPkBQA@mail.gmail.com>
References: <CAA=kHzLbobvFF737zV8Nn+DYrSu9dFommDG=o5ONeoatpPkBQA@mail.gmail.com>
Message-ID: <504C2F2B.7030502@gmail.com>

On 09.09.2012 05:40, Joshua Foshee wrote:
> We have a issue with the call quality of the recorded call to be in a
> faster speed then what the call takes place in. I have took a packet
> capture on the server to replay the audio stream and it sounds just
> fine. Anyone have the same problem with the speed of the audio recording

This happens because of transcoding. If you look carefully, the bad 
audio will be only for certain users' sip phones. Look for phones 
connected to problematic extensions and change in phone settings the 
signaling standard from china to other. Also update FS to latest 1.3.0 
release.

-- 
Mimiko desu.


From yehavi.bourvine at gmail.com  Sun Sep  9 18:31:13 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 9 Sep 2012 17:31:13 +0300
Subject: [Freeswitch-users] effective_callee_id_name behaviour
In-Reply-To: <5048F788.8040005@gmail.com>
References: <CAAtWKeEXaWLfXL89QcwxHCHXWE07up13mgOwUUCytRNpv9EogQ@mail.gmail.com>
	<CAKbxfG_=OZHu7hp9K9MJDobRFCce4Dse4n1BmVubb47tYpFncw@mail.gmail.com>
	<5048F788.8040005@gmail.com>
Message-ID: <CAAtWKeGO61--7h16RQjgZjKzgVCvW5War0WY=M3C0UhbH1TsHg@mail.gmail.com>

Will do once I find out the correct combination...
What Anthony suggested partially works, and I have yet to find why...

                       Thanks, __Yehavi:

2012/9/6 Abaci <abaci64 at gmail.com>

> Yehavi,
> I don't see these variables documented on wiki, can you please document
> them once you get them working.
> see http://wiki.freeswitch.org/wiki/Channel_Variables#Callee_ID_Related
>
> On 9/5/2012 12:04 PM, Anthony Minessale wrote:
> > {origination_callee_id_name='test
> user',origination_callee_id_number=5551212}
> >
> >
> > On Wed, Sep 5, 2012 at 5:56 AM, Yehavi Bourvine
> > <yehavi.bourvine at gmail.com> wrote:
> >> Hello,
> >>
> >>    Our FreeSwitch is connected to a Nortel PBX via E1-SIP gateway
> >> (AudioCodes). This gateway supports the P-Asserted-ID field.
> >>
> >> With a vanilla configuration the following happens when a FS user calls
> a
> >> user on the Nortel:
> >> - During the ringing phase the name of the callee is "outbound call"
> >> - When the other side answers the name of the callee is set to the one
> sent
> >> from the Nortel.
> >>
> >> In order to have the name also during the ringing phase, I set
> calle_id_name
> >> and effective_callee_id_name.
> >> After doing so, the name is not changed after the remote user answers. I
> >> need it to change to the name sent
> >> from the Nortel, as it shows the name of the one who actually answered
> the
> >> phone.
> >>
> >> I also tried adding ignore_display_updates=false, but it didn't change
> the
> >> behaviour.
> >>
> >> Any idea how to do it?
> >>
> >>                          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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sun Sep  9 19:59:22 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 9 Sep 2012 16:59:22 +0100
Subject: [Freeswitch-users] Confusion about sip hangup cause Q850 hangup
 cause and long struggles with a provider.
In-Reply-To: <CAKbxfG-w6N2RvQ5SvAgh0qnCyaSRq=BDnL7MWaERevM4P86k-g@mail.gmail.com>
References: <1407731.17913.1347006256794.JavaMail.root@mail>
	<22952124.18075.1347007177101.JavaMail.root@mail>
	<CAKbxfG-w6N2RvQ5SvAgh0qnCyaSRq=BDnL7MWaERevM4P86k-g@mail.gmail.com>
Message-ID: <CAFiqYumdeJe27MNnLcBUprTwrw53678t-+CY0skbWdmB09KrHQ@mail.gmail.com>

Shouldn't the presence of a Reason header in the 408 from the callee
override the default rfc mapping though?

-Steve



On 8 September 2012 22:27, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I think our wiki is less accurate than our code:  Our cause mappings are
> straight from RFC 4497 http://tools.ietf.org/html/rfc4497
>
> Here is an excerpt:
>
>  408 Request timeout                 102 Recovery on timer expiry
>
>  504 Gateway time-out                102 Recovery on timer expiry
>
>
>
>
>
> On Fri, Sep 7, 2012 at 3:39 AM, Sias Mey <sias at cpdata.co.za> wrote:
>>
>> Hi, Im sure this subject has been beaten to death .. but many googles and
>> many email searches hasent really managed to find me something.
>>
>> Im a dev for a small company that writes call center software. Freeswitch
>> was a godsend, thank you.
>>
>> Now .. the confusion.
>>
>> We are getting a lot of what seem to be strange hangup codes from a new
>> provider big fights about loads of failled calls ensued blah blah.. much sip
>> packet logging and manual inspection later.. I found the following.
>>
>> from xmlcdr.
>>
>>
>>     <sip_hangup_disposition>recv_refuse</sip_hangup_disposition>
>>     <sip_term_status>408</sip_term_status>
>>     <proto_specific_hangup_cause>sip%3A408</proto_specific_hangup_cause>
>>     <sip_term_cause>102</sip_term_cause>
>>     <hangup_cause>RECOVERY_ON_TIMER_EXPIRE</hangup_cause>
>>     <hangup_cause_q850>102</hangup_cause_q850>
>>
>> this just to show its the same call
>>     <sip_call_id>0d4d4e76-735a-1230-d2ac-000423b5571b</sip_call_id>
>>
>> and from the sip messages.
>>
>> SIP/2.0 408 Request Timeout
>> Call-ID: 0d4d4e76-735a-1230-d2ac-000423b5571b
>> Reason: Q.850;cause=18;text="no user responding"
>>
>> And according to the very useful wiki page on Q.850 codes 408 should = 18
>> like it does in the providers response.
>>
>> Why then is the q850 hangup cause in the CDR 102? and where does that
>> translation come from.
>>
>> This is a single example but I also have loads and loads where the CDR
>> claims q850 code 18 but the sip messages provide 31 or a range of other
>> codes.
>> I can understand if the q850 code from the sip message is not being read
>> by FS since FS has to be a bit more agnostic than that and in the pas I have
>> almost exclusively worked with direct connections to TDM hardware so my
>> knowledge and understanding of the sip messages is rather limited. But even
>> in that case, shouldent the q850 code in the cdr at least conform to the
>> translation from the wiki page?
>>
>> Oh I am not currently running the latest git release, having some libtiff
>> issues on ubuntu to compile. I will respond to this again if I manage that
>> and it helps matters.
>>
>> Thank you for your time and help,
>> Regards
>> Sias
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 sias at cpdata.co.za  Sun Sep  9 20:40:16 2012
From: sias at cpdata.co.za (Sias Mey)
Date: Sun, 9 Sep 2012 18:40:16 +0200 (SAST)
Subject: [Freeswitch-users] Confusion about sip hangup cause Q850 hangup
 cause and long struggles with a provider.
In-Reply-To: <CAFiqYumdeJe27MNnLcBUprTwrw53678t-+CY0skbWdmB09KrHQ@mail.gmail.com>
Message-ID: <13167169.25101.1347208816104.JavaMail.root@mail>

Actually Anthony your wiki and code are accurate .. the rfc maps the translations different depending on direction.

Q.850 18 = sip 408
sip 408 = Q.850 102

sip 480 = Q.850 18

Is there a way for me to access the reason header from the sip message?

----- Original Message -----
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, 9 September, 2012 5:59:22 PM
Subject: Re: [Freeswitch-users] Confusion about sip hangup cause Q850 hangup cause and long struggles with a provider.

Shouldn't the presence of a Reason header in the 408 from the callee
override the default rfc mapping though?

-Steve



On 8 September 2012 22:27, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I think our wiki is less accurate than our code:  Our cause mappings are
> straight from RFC 4497 http://tools.ietf.org/html/rfc4497
>
> Here is an excerpt:
>
>  408 Request timeout                 102 Recovery on timer expiry
>
>  504 Gateway time-out                102 Recovery on timer expiry
>
>
>
>
>
> On Fri, Sep 7, 2012 at 3:39 AM, Sias Mey <sias at cpdata.co.za> wrote:
>>
>> Hi, Im sure this subject has been beaten to death .. but many googles and
>> many email searches hasent really managed to find me something.
>>
>> Im a dev for a small company that writes call center software. Freeswitch
>> was a godsend, thank you.
>>
>> Now .. the confusion.
>>
>> We are getting a lot of what seem to be strange hangup codes from a new
>> provider big fights about loads of failled calls ensued blah blah.. much sip
>> packet logging and manual inspection later.. I found the following.
>>
>> from xmlcdr.
>>
>>
>>     <sip_hangup_disposition>recv_refuse</sip_hangup_disposition>
>>     <sip_term_status>408</sip_term_status>
>>     <proto_specific_hangup_cause>sip%3A408</proto_specific_hangup_cause>
>>     <sip_term_cause>102</sip_term_cause>
>>     <hangup_cause>RECOVERY_ON_TIMER_EXPIRE</hangup_cause>
>>     <hangup_cause_q850>102</hangup_cause_q850>
>>
>> this just to show its the same call
>>     <sip_call_id>0d4d4e76-735a-1230-d2ac-000423b5571b</sip_call_id>
>>
>> and from the sip messages.
>>
>> SIP/2.0 408 Request Timeout
>> Call-ID: 0d4d4e76-735a-1230-d2ac-000423b5571b
>> Reason: Q.850;cause=18;text="no user responding"
>>
>> And according to the very useful wiki page on Q.850 codes 408 should = 18
>> like it does in the providers response.
>>
>> Why then is the q850 hangup cause in the CDR 102? and where does that
>> translation come from.
>>
>> This is a single example but I also have loads and loads where the CDR
>> claims q850 code 18 but the sip messages provide 31 or a range of other
>> codes.
>> I can understand if the q850 code from the sip message is not being read
>> by FS since FS has to be a bit more agnostic than that and in the pas I have
>> almost exclusively worked with direct connections to TDM hardware so my
>> knowledge and understanding of the sip messages is rather limited. But even
>> in that case, shouldent the q850 code in the cdr at least conform to the
>> translation from the wiki page?
>>
>> Oh I am not currently running the latest git release, having some libtiff
>> issues on ubuntu to compile. I will respond to this again if I manage that
>> and it helps matters.
>>
>> Thank you for your time and help,
>> Regards
>> Sias
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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


From rzheng at gmail.com  Sun Sep  9 11:01:07 2012
From: rzheng at gmail.com (Richard Zheng)
Date: Sat, 8 Sep 2012 21:01:07 -1000
Subject: [Freeswitch-users] map inbound calls to a gateway based on IP
Message-ID: <CACcFbj7rpaXLRsBW6qKqDNs5nk_bzNw=Q6MXTxfmVxQw3XZ3Lg@mail.gmail.com>

Hi,

I know that this has been discussed many times.  Just want to get a clear
answer.  Is it possible to map inbound calls to a particular gateway based
on IP address?  Someone seems to claim that it's working.  Others say no
way.  I tried to put context param in gateway section and no effect.  This
is in sip_profile gateway config.  There were suggestions to just use
context public, then send to its own context in dialplan.  But that's not
all the same.

In asterisk world, the following section is sufficient in sip.conf.  The
system matches IP based SIP traffic first, then look for users with
registration based.

[providerA]
type=peer
qualify=yes
context=provider_a
host=1.2.3.4
disallow=all      ; First disallow all codecs
allow=ulaw      ; Allow codecs in order of preference
fromuser=
dtmfmode=rfc2833
fromdomain=provider_a

Richard
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From mitch.capper at gmail.com  Mon Sep 10 09:10:16 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sun, 9 Sep 2012 22:10:16 -0700
Subject: [Freeswitch-users] Question about Exporting the cacaert.pem for
 client devices
In-Reply-To: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039BFD9@SN2PRD0610MB384.namprd06.prod.outlook.com>
References: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039B9D1@SN2PRD0610MB384.namprd06.prod.outlook.com>
	<CALn233pGtkcGVQ+M2EF1iGV5=rCW-fjASdxyFBGx90BtnKERfw@mail.gmail.com>
	<D582AE1C5F25FE409F6BC8BAB69D9F4F3039BFD9@SN2PRD0610MB384.namprd06.prod.outlook.com>
Message-ID: <CALn233qGa2bTnoC+AozojqxiFbPW-ijkzFC2qC6XbEyry80MRg@mail.gmail.com>

Yes the contents of that file you do not need to convert it to a .crt
it is good to go.

~Mitch

On Sat, Sep 8, 2012 at 9:01 AM, Andrew Carrega <acarrega at vartel.com> wrote:
> Hi Mitch,
> thanks for your help. You say below copy this ca.cert.pem file. "that's where I am stuck" are you suggesting I can copy and paste the contents from this file or "copy the file" off of the freeswitch server. If you are saying copy the ca.cert.pem file off the server that is where I am stuck. I can't get to that directory /usr/local/freeswitch/conf/ssl/ to copy the file.  I can view the contents of the ca.cert.pem file with the
>
> openssl x509 -noout -inform pem -text -in /usr/local/freeswitch/conf/ssl/CA/ca.cert.pem comand but that is all I am able to do at the moment.
>
> ________________________________________
> From: Mitch Capper [mitch.capper at gmail.com]
> Sent: Friday, September 07, 2012 10:08 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Question about Exporting the cacaert.pem for    client devices
>
> Sure so the cafile.pem should only contain a "BEGIN CERTIFICATE" and
> "END CERTIFICATE" block no PRIVATE KEY.   You can copy this file and
> most clients will expect a .crt file, you can just rename it from
> cafile.pem to ca.crt.   As for loading it into a specific client that
> will depend on the sip client.  If its a softphone it may trust any CA
> installed in the windows certificate store, in which case you can
> double click and open the .crt file in windows and just import it.
> Otherwise search for the phone and "server certificate" or "ca
> certificate" and import and it should have details.
>
>
> ~Mitch
>
> On Fri, Sep 7, 2012 at 12:26 PM, Andrew Carrega <acarrega at vartel.com> wrote:
>> I followed the Freeswitch wiki for enabling tls & srtp on Freeswitch. I have
>> it enabled on my internal and external profiles and both profiles are
>> starting up just fine.
>>
>> I can review my certificate details with the command:
>>
>> openssl x509 -noout -inform pem -text -in
>> /usr/local/freeswitch/conf/ssl/agent.pem
>>
>>
>>
>> I am not at this section of the wiki where it says the clients should have
>> at least the CA root certificate.
>>
>> Clients should all have at least the CA root certificate installed onto them
>> in order to ensure security. Without enabling chain verification (that the
>> server certificate was issued by the approved CA) a MITM attack is possible
>> against a client. The CA certificate is the conf/ssl/cafile.pem it contains
>> only a certificate and clients use it to ensure the server certificate is
>> issued by the CA.
>>
>>
>>
>> Where I am stuck is understanding how to export or download the cacert.pem
>> from the server? I seem to don?t understand the process or tools to use and
>> I can?t seem to access /usr/local/freeswitch/conf/ssl  directory or the
>> /usr/local/freeswitch/conf/ssl/CA from root.
>>
>>
>>
>> Any help is appreciated.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ztillma at gmail.com  Mon Sep 10 07:10:44 2012
From: ztillma at gmail.com (Zaid Tillma)
Date: Sun, 9 Sep 2012 23:10:44 -0400
Subject: [Freeswitch-users] simultaneous calls
Message-ID: <20120910031044.GE15903@ztillma.com>

My goal is to have two phone announce a recording (in a wav file). I have accomplished goal in the following manner:

I have an extension configured like this:

<extension name="wavs">
    <condition field="destination_number" expression="^2009$">
      <action application="playback" data="/path/to/your.wav"/>
    </condition>
  </extension>

When you call, you get an audio playback.

I am trying to originate a call simultaneously to two separate phone (i.e want to two phone to play audio) at the same time, and  I am doing it like this (and it works):

/usr/local/freeswitch/bin/fs_cli -x 'originate {ignore_early_media=true,sip_auto_answer=true}user/1000  2009'
/usr/local/freeswitch/bin/fs_cli -x 'originate {ignore_early_media=true,sip_auto_answer=true}user/1001  2009'

There's got to be a better way to do this way without having two seperate originates.. I did try the Enterprise originate, but only one phone answered... Not sure if there was a mistake in my Enterprise originate implementation.

Please let me know if there is a more elegant way to do this..



From TamurlangClan at gmail.com  Mon Sep 10 10:09:34 2012
From: TamurlangClan at gmail.com (TamurlangClan)
Date: Sun, 9 Sep 2012 23:09:34 -0700 (PDT)
Subject: [Freeswitch-users] Easy Information Needed For Placement Of POP To
 Avoid Latency
Message-ID: <34410404.post@talk.nabble.com>


Hi,

I am weak in network architecture and need help in placement of POP. Please
see below:

1) Is it must to have a POP close to where your subsicriber base. For
Example: If my user base is in Australia as well, Will they be affected if
their calls are routed through POP in New York as it might cause Latency and
packet delay thereby dropping call quality considerably? 

2) I have seen some termination providers who have indirect route. Does this
mean if i choose them my calls will be routed through multiple switches at
multiple locations which will give poor quality voice and also high Latency
and Packet Delay? So is it better choose direct routes?

Please guide on this. Thanks in advance.
-- 
View this message in context: http://old.nabble.com/Easy-Information-Needed-For-Placement-Of-POP-To-Avoid-Latency-tp34410404p34410404.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From gabe at gundy.org  Mon Sep 10 11:47:50 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 10 Sep 2012 01:47:50 -0600
Subject: [Freeswitch-users] simultaneous calls
In-Reply-To: <20120910031044.GE15903@ztillma.com>
References: <20120910031044.GE15903@ztillma.com>
Message-ID: <CAMwZ-tLvETVM5VtgovcX-CXBfVSdzoN3bVu3QYEwvqsTzxaM8g@mail.gmail.com>

On Sun, Sep 9, 2012 at 9:10 PM, Zaid Tillma <ztillma at gmail.com> wrote:
> I am trying to originate a call simultaneously to two separate phone (i.e want to two phone to play audio) at the same time, and  I am doing it like this (and it works):

What if they don't answer at the same time?


Gabe


From acarrega at vartel.com  Mon Sep 10 15:48:03 2012
From: acarrega at vartel.com (Andrew Carrega)
Date: Mon, 10 Sep 2012 11:48:03 +0000
Subject: [Freeswitch-users] map inbound calls to a gateway based on IP
In-Reply-To: <CACcFbj7rpaXLRsBW6qKqDNs5nk_bzNw=Q6MXTxfmVxQw3XZ3Lg@mail.gmail.com>
References: <CACcFbj7rpaXLRsBW6qKqDNs5nk_bzNw=Q6MXTxfmVxQw3XZ3Lg@mail.gmail.com>
Message-ID: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039CBE9@SN2PRD0610MB384.namprd06.prod.outlook.com>



sure. This will work. so if the "sip_contact _host= IP address then bridge



  condition       ${sip_contact_host}     XX.XX.XX.XX           <http://208.75.228.154/app/dialplan/dialplan_details_edit.php?id=134acbb2-75a8-4e8a-89d1-fa32e730e118&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>  <http://208.75.228.154/app/dialplan/dialplan_details_delete.php?id=134acbb2-75a8-4e8a-89d1-fa32e730e118&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>
  condition       context         public                <http://208.75.228.154/app/dialplan/dialplan_details_edit.php?id=38e3a52a-7ac7-45bb-a1e6-512afde61cb9&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>  <http://208.75.228.154/app/dialplan/dialplan_details_delete.php?id=38e3a52a-7ac7-45bb-a1e6-512afde61cb9&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>
  condition       destination_number      ^(\d{10})$            <http://208.75.228.154/app/dialplan/dialplan_details_edit.php?id=7f154265-0f95-4b6c-be75-e2a932ae7307&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>  <http://208.75.228.154/app/dialplan/dialplan_details_delete.php?id=7f154265-0f95-4b6c-be75-e2a932ae7307&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>
  action          set     hangup_after_bridge=true              <http://208.75.228.154/app/dialplan/dialplan_details_edit.php?id=53b85bcc-33e6-4570-b645-6d30585deeee&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>  <http://208.75.228.154/app/dialplan/dialplan_details_delete.php?id=53b85bcc-33e6-4570-b645-6d30585deeee&id2=43aec539-2cf6-47c9-bb74-587a31a8e69f>
  action          bridge          sofia/gateway/GatewayName/$1

________________________________
From: Richard Zheng [rzheng at gmail.com]
Sent: Sunday, September 09, 2012 3:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] map inbound calls to a gateway based on IP

Hi,

I know that this has been discussed many times.  Just want to get a clear answer.  Is it possible to map inbound calls to a particular gateway based on IP address?  Someone seems to claim that it's working.  Others say no way.  I tried to put context param in gateway section and no effect.  This is in sip_profile gateway config.  There were suggestions to just use context public, then send to its own context in dialplan.  But that's not all the same.

In asterisk world, the following section is sufficient in sip.conf.  The system matches IP based SIP traffic first, then look for users with registration based.

[providerA]
type=peer
qualify=yes
context=provider_a
host=1.2.3.4
disallow=all      ; First disallow all codecs
allow=ulaw      ; Allow codecs in order of preference
fromuser=
dtmfmode=rfc2833
fromdomain=provider_a

Richard
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From jaybinks at gmail.com  Mon Sep 10 15:58:57 2012
From: jaybinks at gmail.com (jay binks)
Date: Mon, 10 Sep 2012 21:58:57 +1000
Subject: [Freeswitch-users] Easy Information Needed For Placement Of POP
 To Avoid Latency
In-Reply-To: <34410404.post@talk.nabble.com>
References: <34410404.post@talk.nabble.com>
Message-ID: <CAO47HpcLOz3hL6gv=6PcM+YnfzJXss-eSm22vYh5KBMmaB6k8Q@mail.gmail.com>

you are 100% correct... you do not want your traffic routed through the US
if your users are in Australia.

also you are always better to choose a provider with the shortest path to
you ( most direct )
because of latency, packet loss and jitter.

<shameless plug>
if your users truly are in Australia you should select the BEST SIP
trunking provider in Australia.
http://netsip.com.au/

We are also supporters and users of Freeswitch, so you also help the
community by signing up with us :)
</shameless plug>





On 10 September 2012 16:09, TamurlangClan <TamurlangClan at gmail.com> wrote:

>
> Hi,
>
> I am weak in network architecture and need help in placement of POP. Please
> see below:
>
> 1) Is it must to have a POP close to where your subsicriber base. For
> Example: If my user base is in Australia as well, Will they be affected if
> their calls are routed through POP in New York as it might cause Latency
> and
> packet delay thereby dropping call quality considerably?
>
> 2) I have seen some termination providers who have indirect route. Does
> this
> mean if i choose them my calls will be routed through multiple switches at
> multiple locations which will give poor quality voice and also high Latency
> and Packet Delay? So is it better choose direct routes?
>
> Please guide on this. Thanks in advance.
> --
> View this message in context:
> http://old.nabble.com/Easy-Information-Needed-For-Placement-Of-POP-To-Avoid-Latency-tp34410404p34410404.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
>



-- 
Sincerely

Jay
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From acarrega at vartel.com  Mon Sep 10 17:56:33 2012
From: acarrega at vartel.com (Andrew Carrega)
Date: Mon, 10 Sep 2012 13:56:33 +0000
Subject: [Freeswitch-users] Question about Exporting the cacaert.pem
	for	client devices
In-Reply-To: <CALn233qGa2bTnoC+AozojqxiFbPW-ijkzFC2qC6XbEyry80MRg@mail.gmail.com>
References: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039B9D1@SN2PRD0610MB384.namprd06.prod.outlook.com>
	<CALn233pGtkcGVQ+M2EF1iGV5=rCW-fjASdxyFBGx90BtnKERfw@mail.gmail.com>
	<D582AE1C5F25FE409F6BC8BAB69D9F4F3039BFD9@SN2PRD0610MB384.namprd06.prod.outlook.com>
	<CALn233qGa2bTnoC+AozojqxiFbPW-ijkzFC2qC6XbEyry80MRg@mail.gmail.com>
Message-ID: <D582AE1C5F25FE409F6BC8BAB69D9F4F3039CCF6@SN2PRD0610MB384.namprd06.prod.outlook.com>

Mitch-Is this the command to get the contents for that cafile.pem ??

openssl x509 -noout -inform pem -text -in /usr/local/freeswitch/conf/ssl/cafile.pem 

p.s. I ran this command and copies the output using Notepad and named it cafile.crt( even tries cafile.pem). Tried doubleclicking it in File Explorer to install it but I get an error (Invalid Public Key Security Object File) "this file is invalid for use as the following: Security Certificate. 

Any thoughts.

-----Original Message-----
From: Mitch Capper [mailto:mitch.capper at gmail.com] 
Sent: Monday, September 10, 2012 1:10 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Question about Exporting the cacaert.pem for client devices

Yes the contents of that file you do not need to convert it to a .crt it is good to go.

~Mitch

On Sat, Sep 8, 2012 at 9:01 AM, Andrew Carrega <acarrega at vartel.com> wrote:
> Hi Mitch,
> thanks for your help. You say below copy this ca.cert.pem file. 
> "that's where I am stuck" are you suggesting I can copy and paste the 
> contents from this file or "copy the file" off of the freeswitch 
> server. If you are saying copy the ca.cert.pem file off the server 
> that is where I am stuck. I can't get to that directory 
> /usr/local/freeswitch/conf/ssl/ to copy the file.  I can view the 
> contents of the ca.cert.pem file with the
>
> openssl x509 -noout -inform pem -text -in /usr/local/freeswitch/conf/ssl/CA/ca.cert.pem comand but that is all I am able to do at the moment.
>
> ________________________________________
> From: Mitch Capper [mitch.capper at gmail.com]
> Sent: Friday, September 07, 2012 10:08 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Question about Exporting the cacaert.pem for    client devices
>
> Sure so the cafile.pem should only contain a "BEGIN CERTIFICATE" and
> "END CERTIFICATE" block no PRIVATE KEY.   You can copy this file and
> most clients will expect a .crt file, you can just rename it from
> cafile.pem to ca.crt.   As for loading it into a specific client that
> will depend on the sip client.  If its a softphone it may trust any CA 
> installed in the windows certificate store, in which case you can 
> double click and open the .crt file in windows and just import it.
> Otherwise search for the phone and "server certificate" or "ca 
> certificate" and import and it should have details.
>
>
> ~Mitch
>
> On Fri, Sep 7, 2012 at 12:26 PM, Andrew Carrega <acarrega at vartel.com> wrote:
>> I followed the Freeswitch wiki for enabling tls & srtp on Freeswitch. 
>> I have it enabled on my internal and external profiles and both 
>> profiles are starting up just fine.
>>
>> I can review my certificate details with the command:
>>
>> openssl x509 -noout -inform pem -text -in 
>> /usr/local/freeswitch/conf/ssl/agent.pem
>>
>>
>>
>> I am not at this section of the wiki where it says the clients should 
>> have at least the CA root certificate.
>>
>> Clients should all have at least the CA root certificate installed 
>> onto them in order to ensure security. Without enabling chain 
>> verification (that the server certificate was issued by the approved 
>> CA) a MITM attack is possible against a client. The CA certificate is 
>> the conf/ssl/cafile.pem it contains only a certificate and clients 
>> use it to ensure the server certificate is issued by the CA.
>>
>>
>>
>> Where I am stuck is understanding how to export or download the 
>> cacert.pem from the server? I seem to don't understand the process or 
>> tools to use and I can't seem to access 
>> /usr/local/freeswitch/conf/ssl  directory or the /usr/local/freeswitch/conf/ssl/CA from root.
>>
>>
>>
>> Any help is appreciated.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _____________________________________________________________________
>> ____ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users 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





From bclark at grasshopper.com  Mon Sep 10 18:32:57 2012
From: bclark at grasshopper.com (Brett Clark - Grasshopper)
Date: Mon, 10 Sep 2012 09:32:57 -0500
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <CAD-qRkjgK9rGFmznjVxM-CiCFFgYKBSTwOR5RwHevBAfaaviQA@mail.gmail.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
	<CAD-qRkjgK9rGFmznjVxM-CiCFFgYKBSTwOR5RwHevBAfaaviQA@mail.gmail.com>
Message-ID: <A90437D5515AF341A7942C6446201F3E07154E5D94@DFW1MBX20.mex07a.mlsrvr.com>

Hey Chris,

Could you explain what you mean by clustering FS?  I get the idea of clustering, in general, but I what do you mean in this context?

Thanks!
Brett


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Friday, September 07, 2012 12:14 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] best FS interface for development

For arbitrarily large and complex systems, I'd keep FreeSWITCH as dumb as possible so that you can easily cluster them.  So, modules that allow external control (mod_event_socket, mod_erlang_event, mod_httapi, mod_xml_curl) are all good choices.

Event socket is pretty popular; Plivo and Adhearsion both use it.  The 2600hz guys use mod_erlang_event.  And I've heard of plenty of people that use mod_xml_curl.

Chris


On Fri, Sep 7, 2012 at 9:43 AM, Brett Clark - Grasshopper <bclark at grasshopper.com<mailto:bclark at grasshopper.com>> wrote:
There seems to be at least 4 general ways for customizing FS:
- Use the event socket and build an app to interface with it
- Use mod lua, or similar, to develop in a particular language
- Develop your own 'module' to interface directly with the core
- Use the XML interface to implement IVR, PBX, and voicemail type functionality and all the rest.

How prevalent is each approach in the community?  It seems like everyone is doing a little of both, but maybe someone has a better handle of what the actual ratios are?

Which is best suited for an arbitrarily large and complex application?  I realize this isn't a simple question-what I want is to understand the most evolved and featureful way to interface with FS?  I don't want to adopt an approach which isn't being actively maintained or is missing features.  As new stuff is added to FS, which approach will allow me to adopt those new features most easily?

Thanks!
Brett

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From daveh at beachdognet.com  Mon Sep 10 18:51:53 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Mon, 10 Sep 2012 10:51:53 -0400
Subject: [Freeswitch-users] stereo recording problem - tracks out of sync
Message-ID: <k2kuq6$qje$1@ger.gmane.org>

I have an application where I create a bridging conference that 
outdials a B party, and then I record the stream between the conference 
and the B party.  However, the two channels of the resulting stereo 
recording are not in sync: for instance, if I call into an IVR and 
barge into a prompt by pressing a dtmf, on the recording the dtmf tone 
appears several seconds after the barged-in prompt stopped.  Its as if 
the stream from the conference to the B leg has been shifted in time so 
it no longer matches up properly with the conference mix.  I can't tell 
if this variance is constant or gets worse as the conference goes on.  
Has anyone a notion of what might be going on here?  Here is my 
dialplan:

    <extension name="6174973895" continue="false">
      <condition field="source" expression="mod_sofia"/>
      <condition field="destination_number" expression="6174973895">
                <action application="ring_ready"/>
                <action application="answer"/>
                <action application="start_dtmf"/>
                <action application="set" 
data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
                <action application="export" 
data="outfile=$${base_dir}/tmp/${call_id}.wav"/>

                <action application="lua" data="demo_call2.lua"/>

                <action application="export" data="RECORD_STEREO=true"/>
                <action application="export" 
data="nolocal:execute_on_media=record_session ${outfile}"/>

                <action application="export" 
data="nolocal:api_on_answer=conference ${call_id} play 
misc/this-call-may-be-recorded.wav async"/>
                <action application="conference" data="bridge: 
${call_id}@simple:{origination_caller_id_number=${caller_id_number}}sofia/normal_customer/${outdial}@${egress_gateway}"/> 

        </condition>
    </extension>




From lists at kavun.ch  Mon Sep 10 19:17:43 2012
From: lists at kavun.ch (Emrah)
Date: Mon, 10 Sep 2012 11:17:43 -0400
Subject: [Freeswitch-users] Enter and exit sound in a conference with
	wait-mod flag
Message-ID: <B497E97F-3B8E-4A62-B06C-11D0840CE4FC@kavun.ch>

Morning guys,

I noticed something that sounds inconsistent and would appreciate if you could confirm it.
I have a conference profile with the wait-mod flag set on.
Upon entering the conference as a user, the MOH stops for current participants, the enter sound is played and MOH resumes. This is more or less a 1 second pause.
However, when I exit the call as a user, the exit sound is played mixed with the MOH with no pause whatsoever.

How can we manage the pause there?

I would ideally want to be able to pause / resume the MOH in a conference in order to allow the enter / exit sound played along with the name of the caller, without the distraction of the MOH. I announce callers when they enter and exit the conference. 

Any good hints would be appreciated.

Best,
Emrah

From mkdutchman at gmail.com  Mon Sep 10 12:35:25 2012
From: mkdutchman at gmail.com (Melvin King)
Date: Mon, 10 Sep 2012 04:35:25 -0400
Subject: [Freeswitch-users] Odd SIP user registration problem
In-Reply-To: <-338675922493849483@unknownmsgid>
References: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>
	<8333860497797977792@unknownmsgid>
	<D582AE1C5F25FE409F6BC8BAB69D9F4F3039C108@SN2PRD0610MB384.namprd06.prod.outlook.com>
	<CALg5agZU4xXa3_n71OFPi0EqQDMOfw+ykYm-hwyiNXFMicGObQ@mail.gmail.com>
	<-338675922493849483@unknownmsgid>
Message-ID: <CALg5agbGTYH9Ex6pw3NMaC5i70rECzbwcxEPfb1HBQNuHwXRSQ@mail.gmail.com>

They are using just the standard username and password

Mel

On Sun, Sep 9, 2012 at 1:18 AM, Chris Ferreira <chris at gonumina.com> wrote:

> Where are your phones connecting from? Are they using domains in their
> authentication?
>
>
>
> ___________________
> Mobile Reply
>
> On Sep 8, 2012, at 10:12 PM, Melvin King <mkdutchman at gmail.com> wrote:
>
> 23.21.191.100 is the static address of the WAN side of the network that
> the freeswitch host is connected to.
>
> STUN is turned off.
>
> 23.21.191.100 is set as the external_rtp_ip and external_sip_ip
>
> Mel
>
> On Sat, Sep 8, 2012 at 3:41 PM, Andrew Carrega <acarrega at vartel.com>wrote:
>
>>  What do you have set for your external_rtp_ip and external_sip_ip??
>> ------------------------------
>> *From:* Chris Ferreira [chris at gonumina.com]
>> *Sent:* Saturday, September 08, 2012 11:55 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Odd SIP user registration problem
>>
>>   See what that IP is 23.21.191.100.
>>
>>
>>  It looks like this is a STUN Server issue on the Sip Phones.
>>
>> ___________________
>> Mobile Reply
>>
>> On Sep 8, 2012, at 9:16 AM, Melvin King <mkdutchman at gmail.com> wrote:
>>
>>   Hello,
>> I've been beating my brains out for a while with this problem, I think I
>> might need some help. I've been trying to set it up as a trimmed down to
>> the max carrier grade SIP PBX.
>>
>> Here's my setup
>>
>> I'm using a amazon ec2 64 bit image with a public static IP for the host.
>> (Inside a NAT)
>>
>> ports opened from the outside are
>> TCP 5060
>> UDP 16384 - 32768
>>
>> FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2;
>> git at commit e97da8e20a
>>
>> This is the relevant error message that I am getting, I cannot register
>> any users at all, even though they're in the directory
>>
>> 2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [
>> 1000 at 23.21.191.100]
>> 2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [
>> 1000 at 23.21.191.100]
>> You must define a domain called '23.21.191.100' in your directory and add
>> a user with the id="1000" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>> 2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [
>> 1000 at 23.21.191.100]
>> 2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [
>> 1000 at 23.21.191.100]
>> 2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded
>> flowroute with code 200 - count -1/1/1, state UP
>> 2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [
>> 1000 at 23.21.191.100]
>> You must define a domain called '23.21.191.100' in your directory and add
>> a user with the id="1000" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>> 2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [
>> 1000 at 23.21.191.100]
>>
>>
>> Any insight would be greatly appreciated
>>
>> Mel
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ztillma at gmail.com  Mon Sep 10 15:14:26 2012
From: ztillma at gmail.com (Zaid Tillma)
Date: Mon, 10 Sep 2012 07:14:26 -0400
Subject: [Freeswitch-users] simultaneous calls
Message-ID: <20120910111426.GF15903@ztillma.com>

On Sun, Sep 10, 2012 at 11:10 AM, Gabriel Gunderson <gabe at gundy.org> wrote:
> What if they don't answer at the same time?

Both of the extensions are set to auto-answer. In case one of them does not answer because it is being used, play on the non-busy extension. Once the busy extension gets un-busy, play the remaining audio on it (i.e does not have to start from the begining). I understand that this will lead to the case where one extension does not play the audio at all.

Now in case both are busy, wait till one of them gets unbusy and then play. Then the plan listed above applies.

FYI - I get a daily digest, and I am replying to Gabe's email by copy & paste from the digest. If this does not work, I'd appreciate someone letting me know how to post correctly if all you get is the daily digest.

Thanks
Z



From chris at gonumina.com  Mon Sep 10 19:33:51 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Mon, 10 Sep 2012 23:33:51 +0800
Subject: [Freeswitch-users] Odd SIP user registration problem
In-Reply-To: <CALg5agbGTYH9Ex6pw3NMaC5i70rECzbwcxEPfb1HBQNuHwXRSQ@mail.gmail.com>
References: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>
	<8333860497797977792@unknownmsgid>
	<D582AE1C5F25FE409F6BC8BAB69D9F4F3039C108@SN2PRD0610MB384.namprd06.prod.outlook.com>
	<CALg5agZU4xXa3_n71OFPi0EqQDMOfw+ykYm-hwyiNXFMicGObQ@mail.gmail.com>
	<-338675922493849483@unknownmsgid>
	<CALg5agbGTYH9Ex6pw3NMaC5i70rECzbwcxEPfb1HBQNuHwXRSQ@mail.gmail.com>
Message-ID: <CAEuuoeOK6Hc3u4ccygi3GBp7HWoN9c4=atbEzkQzNfco=a=jjg@mail.gmail.com>

Try enabling Stun on a phone and see if it works. Try s free Soft Phone
like Zoiper that has stun already setup if you want.





On Mon, Sep 10, 2012 at 4:35 PM, Melvin King <mkdutchman at gmail.com> wrote:

> They are using just the standard username and password
>
> Mel
>
>
> On Sun, Sep 9, 2012 at 1:18 AM, Chris Ferreira <chris at gonumina.com> wrote:
>
>> Where are your phones connecting from? Are they using domains in their
>> authentication?
>>
>>
>>
>> ___________________
>> Mobile Reply
>>
>> On Sep 8, 2012, at 10:12 PM, Melvin King <mkdutchman at gmail.com> wrote:
>>
>> 23.21.191.100 is the static address of the WAN side of the network that
>> the freeswitch host is connected to.
>>
>> STUN is turned off.
>>
>> 23.21.191.100 is set as the external_rtp_ip and external_sip_ip
>>
>> Mel
>>
>> On Sat, Sep 8, 2012 at 3:41 PM, Andrew Carrega <acarrega at vartel.com>wrote:
>>
>>>  What do you have set for your external_rtp_ip and external_sip_ip??
>>> ------------------------------
>>> *From:* Chris Ferreira [chris at gonumina.com]
>>> *Sent:* Saturday, September 08, 2012 11:55 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Odd SIP user registration problem
>>>
>>>   See what that IP is 23.21.191.100.
>>>
>>>
>>>  It looks like this is a STUN Server issue on the Sip Phones.
>>>
>>> ___________________
>>> Mobile Reply
>>>
>>> On Sep 8, 2012, at 9:16 AM, Melvin King <mkdutchman at gmail.com> wrote:
>>>
>>>   Hello,
>>> I've been beating my brains out for a while with this problem, I think I
>>> might need some help. I've been trying to set it up as a trimmed down to
>>> the max carrier grade SIP PBX.
>>>
>>> Here's my setup
>>>
>>> I'm using a amazon ec2 64 bit image with a public static IP for the
>>> host. (Inside a NAT)
>>>
>>> ports opened from the outside are
>>> TCP 5060
>>> UDP 16384 - 32768
>>>
>>> FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2;
>>> git at commit e97da8e20a
>>>
>>> This is the relevant error message that I am getting, I cannot register
>>> any users at all, even though they're in the directory
>>>
>>> 2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [
>>> 1000 at 23.21.191.100]
>>> 2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [
>>> 1000 at 23.21.191.100]
>>> You must define a domain called '23.21.191.100' in your directory and
>>> add a user with the id="1000" attribute
>>> and you must configure your device to use the proper domain in it's
>>> authentication credentials.
>>> 2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [
>>> 1000 at 23.21.191.100]
>>> 2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [
>>> 1000 at 23.21.191.100]
>>> 2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded
>>> flowroute with code 200 - count -1/1/1, state UP
>>> 2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [
>>> 1000 at 23.21.191.100]
>>> You must define a domain called '23.21.191.100' in your directory and
>>> add a user with the id="1000" attribute
>>> and you must configure your device to use the proper domain in it's
>>> authentication credentials.
>>> 2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [
>>> 1000 at 23.21.191.100]
>>>
>>>
>>> Any insight would be greatly appreciated
>>>
>>> Mel
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris.chen2004 at gmail.com  Mon Sep 10 19:38:10 2012
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 10 Sep 2012 11:38:10 -0400
Subject: [Freeswitch-users] Odd SIP user registration problem
In-Reply-To: <CALg5agbGTYH9Ex6pw3NMaC5i70rECzbwcxEPfb1HBQNuHwXRSQ@mail.gmail.com>
References: <CALg5agb1K6Y0kTaB1J0vOCkE4fRasE0rZ8GzPecEGTO-nRHzSA@mail.gmail.com>
	<8333860497797977792@unknownmsgid>
	<D582AE1C5F25FE409F6BC8BAB69D9F4F3039C108@SN2PRD0610MB384.namprd06.prod.outlook.com>
	<CALg5agZU4xXa3_n71OFPi0EqQDMOfw+ykYm-hwyiNXFMicGObQ@mail.gmail.com>
	<-338675922493849483@unknownmsgid>
	<CALg5agbGTYH9Ex6pw3NMaC5i70rECzbwcxEPfb1HBQNuHwXRSQ@mail.gmail.com>
Message-ID: <CAFEMH7MKD8T0ZX7NFeJYab9MSjkw9UxWMd9P9yVpap3geq3K9Q@mail.gmail.com>

Hi Mel,
unless you have something

<domain name="23.21.191.100">

in your /usr/local/freeswitch/conf/directory/default.xml

you will keep having the same error.

Hope this directs you to the right track.

Thanks,
Chris

On Mon, Sep 10, 2012 at 4:35 AM, Melvin King <mkdutchman at gmail.com> wrote:

> They are using just the standard username and password
>
> Mel
>
>
> On Sun, Sep 9, 2012 at 1:18 AM, Chris Ferreira <chris at gonumina.com> wrote:
>
>> Where are your phones connecting from? Are they using domains in their
>> authentication?
>>
>>
>>
>> ___________________
>> Mobile Reply
>>
>> On Sep 8, 2012, at 10:12 PM, Melvin King <mkdutchman at gmail.com> wrote:
>>
>> 23.21.191.100 is the static address of the WAN side of the network that
>> the freeswitch host is connected to.
>>
>> STUN is turned off.
>>
>> 23.21.191.100 is set as the external_rtp_ip and external_sip_ip
>>
>> Mel
>>
>> On Sat, Sep 8, 2012 at 3:41 PM, Andrew Carrega <acarrega at vartel.com>wrote:
>>
>>>  What do you have set for your external_rtp_ip and external_sip_ip??
>>> ------------------------------
>>> *From:* Chris Ferreira [chris at gonumina.com]
>>> *Sent:* Saturday, September 08, 2012 11:55 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Odd SIP user registration problem
>>>
>>>   See what that IP is 23.21.191.100.
>>>
>>>
>>>  It looks like this is a STUN Server issue on the Sip Phones.
>>>
>>> ___________________
>>> Mobile Reply
>>>
>>> On Sep 8, 2012, at 9:16 AM, Melvin King <mkdutchman at gmail.com> wrote:
>>>
>>>   Hello,
>>> I've been beating my brains out for a while with this problem, I think I
>>> might need some help. I've been trying to set it up as a trimmed down to
>>> the max carrier grade SIP PBX.
>>>
>>> Here's my setup
>>>
>>> I'm using a amazon ec2 64 bit image with a public static IP for the
>>> host. (Inside a NAT)
>>>
>>> ports opened from the outside are
>>> TCP 5060
>>> UDP 16384 - 32768
>>>
>>> FreeSWITCH Version 1.2.0-rc2+git~20120731T213556Z~e97da8e20a (1.2.0-rc2;
>>> git at commit e97da8e20a
>>>
>>> This is the relevant error message that I am getting, I cannot register
>>> any users at all, even though they're in the directory
>>>
>>> 2012-09-08 10:02:13.716511 [DEBUG] sofia_reg.c:1470 Send challenge for [
>>> 1000 at 23.21.191.100]
>>> 2012-09-08 10:02:13.796513 [WARNING] sofia_reg.c:2442 Can't find user [
>>> 1000 at 23.21.191.100]
>>> You must define a domain called '23.21.191.100' in your directory and
>>> add a user with the id="1000" attribute
>>> and you must configure your device to use the proper domain in it's
>>> authentication credentials.
>>> 2012-09-08 10:02:13.796513 [DEBUG] sofia_reg.c:1414 Send forbidden for [
>>> 1000 at 23.21.191.100]
>>> 2012-09-08 10:02:13.896514 [DEBUG] sofia_reg.c:1470 Send challenge for [
>>> 1000 at 23.21.191.100]
>>> 2012-09-08 10:02:13.976510 [WARNING] sofia.c:5199 Ping succeeded
>>> flowroute with code 200 - count -1/1/1, state UP
>>> 2012-09-08 10:02:13.996519 [WARNING] sofia_reg.c:2442 Can't find user [
>>> 1000 at 23.21.191.100]
>>> You must define a domain called '23.21.191.100' in your directory and
>>> add a user with the id="1000" attribute
>>> and you must configure your device to use the proper domain in it's
>>> authentication credentials.
>>> 2012-09-08 10:02:13.996519 [DEBUG] sofia_reg.c:1414 Send forbidden for [
>>> 1000 at 23.21.191.100]
>>>
>>>
>>> Any insight would be greatly appreciated
>>>
>>> Mel
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Sep 10 19:40:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Sep 2012 08:40:35 -0700
Subject: [Freeswitch-users] simultaneous calls
In-Reply-To: <20120910111426.GF15903@ztillma.com>
References: <20120910111426.GF15903@ztillma.com>
Message-ID: <CAKzWOxXiSmJ4rNAhj_FG0S6d6zALvxL_hWBkLFfhrX1vNcCtDA@mail.gmail.com>

I would look in the vanilla example config at conf/dialplan/default.xml,
specifically at the "mad boss" extensions, which IIRC are 0911 and 0912.
Try them out and see what you get.
-MC

P.S. - If all you get is the daily digest then your replies will not be
part of the conversation thread on Mailman. That's one of the drawbacks to
getting digest vs. all messages.

On Mon, Sep 10, 2012 at 4:14 AM, Zaid Tillma <ztillma at gmail.com> wrote:

> On Sun, Sep 10, 2012 at 11:10 AM, Gabriel Gunderson <gabe at gundy.org>
> wrote:
> > What if they don't answer at the same time?
>
> Both of the extensions are set to auto-answer. In case one of them does
> not answer because it is being used, play on the non-busy extension. Once
> the busy extension gets un-busy, play the remaining audio on it (i.e does
> not have to start from the begining). I understand that this will lead to
> the case where one extension does not play the audio at all.
>
> Now in case both are busy, wait till one of them gets unbusy and then
> play. Then the plan listed above applies.
>
> FYI - I get a daily digest, and I am replying to Gabe's email by copy &
> paste from the digest. If this does not work, I'd appreciate someone
> letting me know how to post correctly if all you get is the daily digest.
>
> Thanks
> Z
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vipkilla at gmail.com  Mon Sep 10 19:52:42 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 10 Sep 2012 11:52:42 -0400
Subject: [Freeswitch-users] FSClient and CELT codec version
Message-ID: <CAC-LwPMbLzi--HFOtE2Q3ziqB3o5q-sC+Y2ETK3vphB+8f2KMQ@mail.gmail.com>

I'm trying to connect to a FS conference with FSClient using the CELT
codec, unfortunately, it is not working due to some codec error.
Im wondering if the problem is the difference in CELT versions. My
FreeSWITCH server is running celt-0.10.0.
What is FSClient running?

Thanks.


From anthony.minessale at gmail.com  Mon Sep 10 20:07:57 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 Sep 2012 11:07:57 -0500
Subject: [Freeswitch-users] Confusion about sip hangup cause Q850 hangup
 cause and long struggles with a provider.
In-Reply-To: <13167169.25101.1347208816104.JavaMail.root@mail>
References: <CAFiqYumdeJe27MNnLcBUprTwrw53678t-+CY0skbWdmB09KrHQ@mail.gmail.com>
	<13167169.25101.1347208816104.JavaMail.root@mail>
Message-ID: <CAKbxfG9C8g-2q6VTFb+ZK-aOz_A+RaXsFjxiF8kmgh_iBgC4QQ@mail.gmail.com>

if it gets a bye with a reason header it puts it in the sip_reason var,
which since the call hangs up now is only accessible in the CDR stage.

On Sun, Sep 9, 2012 at 11:40 AM, Sias Mey <sias at cpdata.co.za> wrote:

> Actually Anthony your wiki and code are accurate .. the rfc maps the
> translations different depending on direction.
>
> Q.850 18 = sip 408
> sip 408 = Q.850 102
>
> sip 480 = Q.850 18
>
> Is there a way for me to access the reason header from the sip message?
>
> ----- Original Message -----
> From: "Steven Ayre" <steveayre at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, 9 September, 2012 5:59:22 PM
> Subject: Re: [Freeswitch-users] Confusion about sip hangup cause Q850
> hangup cause and long struggles with a provider.
>
> Shouldn't the presence of a Reason header in the 408 from the callee
> override the default rfc mapping though?
>
> -Steve
>
>
>
> On 8 September 2012 22:27, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > I think our wiki is less accurate than our code:  Our cause mappings are
> > straight from RFC 4497 http://tools.ietf.org/html/rfc4497
> >
> > Here is an excerpt:
> >
> >  408 Request timeout                 102 Recovery on timer expiry
> >
> >  504 Gateway time-out                102 Recovery on timer expiry
> >
> >
> >
> >
> >
> > On Fri, Sep 7, 2012 at 3:39 AM, Sias Mey <sias at cpdata.co.za> wrote:
> >>
> >> Hi, Im sure this subject has been beaten to death .. but many googles
> and
> >> many email searches hasent really managed to find me something.
> >>
> >> Im a dev for a small company that writes call center software.
> Freeswitch
> >> was a godsend, thank you.
> >>
> >> Now .. the confusion.
> >>
> >> We are getting a lot of what seem to be strange hangup codes from a new
> >> provider big fights about loads of failled calls ensued blah blah..
> much sip
> >> packet logging and manual inspection later.. I found the following.
> >>
> >> from xmlcdr.
> >>
> >>
> >>     <sip_hangup_disposition>recv_refuse</sip_hangup_disposition>
> >>     <sip_term_status>408</sip_term_status>
> >>     <proto_specific_hangup_cause>sip%3A408</proto_specific_hangup_cause>
> >>     <sip_term_cause>102</sip_term_cause>
> >>     <hangup_cause>RECOVERY_ON_TIMER_EXPIRE</hangup_cause>
> >>     <hangup_cause_q850>102</hangup_cause_q850>
> >>
> >> this just to show its the same call
> >>     <sip_call_id>0d4d4e76-735a-1230-d2ac-000423b5571b</sip_call_id>
> >>
> >> and from the sip messages.
> >>
> >> SIP/2.0 408 Request Timeout
> >> Call-ID: 0d4d4e76-735a-1230-d2ac-000423b5571b
> >> Reason: Q.850;cause=18;text="no user responding"
> >>
> >> And according to the very useful wiki page on Q.850 codes 408 should =
> 18
> >> like it does in the providers response.
> >>
> >> Why then is the q850 hangup cause in the CDR 102? and where does that
> >> translation come from.
> >>
> >> This is a single example but I also have loads and loads where the CDR
> >> claims q850 code 18 but the sip messages provide 31 or a range of other
> >> codes.
> >> I can understand if the q850 code from the sip message is not being read
> >> by FS since FS has to be a bit more agnostic than that and in the pas I
> have
> >> almost exclusively worked with direct connections to TDM hardware so my
> >> knowledge and understanding of the sip messages is rather limited. But
> even
> >> in that case, shouldent the q850 code in the cdr at least conform to the
> >> translation from the wiki page?
> >>
> >> Oh I am not currently running the latest git release, having some
> libtiff
> >> issues on ubuntu to compile. I will respond to this again if I manage
> that
> >> and it helps matters.
> >>
> >> Thank you for your time and help,
> >> Regards
> >> Sias
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
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From marketing at cluecon.com  Mon Sep 10 20:11:53 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 10 Sep 2012 09:11:53 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxXVy8xJ6JXD107PEY3ubCES+fhe61prKgbERs22eHpxFQ@mail.gmail.com>

Greetings!

We hope you all had a great week. On our Wednesday conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_05>we discussed
several items as a community. One item of note was how to
handle mailing list posts that are overly broad and reflect a lack of
research on the part of the individual doing the posting. After much
discussion we decided that we would create some online documentation that
helps new ones get their bearings when considering the big picture in
FreeSWITCH. (For example, what are modules and why do we have them?) Thanks
to Dave Kompel for helping to get that started.

We are also pleased to announce that we have started up the
Adopt-a-module<http://wiki.freeswitch.org/wiki/Adopt-a-module>project.
The idea is simple but powerful: community members who are
knowledgeable and enthusiastic about a specific module will volunteer to
"adopt" that module. Adopting a module means doing several things: watching
the mailing list and IRC channel for questions, monitoring the Git
repository for new commits, keeping the module's wiki page up-to-date, and
acting as a bug marshal for any Jira <http://jira.freeswitch.org> tickets
that are opened. We've had several people step up already. Please
visit the list
of modules needing
adoption<http://wiki.freeswitch.org/wiki/Adopt-a-module#Modules_Waiting_for_Adoption>to
see if you there is one that fits your area of expertise. We give a
special note of thanks to Anshel Blum for helping to get this one going.

This week we welcome Ben Langfield and Ben
Klang<http://www.cluecon.com/speaker/111/>who will be discussing the
Adhearsion <http://adhearsion.com/> framework for FreeSWITCH. Adhearsion is
a Ruby-based framework for building telephony applications. You may recall
that Ben Klang joined us at ClueCon 2012 <http://www.cluecon.com> to make
the announcement<http://www.cluecon.com/presentation/news-about-adhearsion-project/>about
Adhearsion being available for FreeSWITCH. We look forward to
learning more about how Adhearsion works with FreeSWITCH.

Let's all have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From krice at freeswitch.org  Mon Sep 10 20:18:23 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 10 Sep 2012 11:18:23 -0500
Subject: [Freeswitch-users] Jira and Fisheye Upgrades in progress
Message-ID: <CC737CFF.4BB6E%krice@freeswitch.org>

Hey Guys,

We?re doing another round up grades to some of the FreeSWITCH Project
support systems... Jira and Fisheye specifically...

So if you suddenly find one or the other not answering via the web, clear
your DNS cache, then try again... We are upgrading hardware and software
concurrently this go around...

K

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From mitch.capper at gmail.com  Mon Sep 10 20:35:32 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 10 Sep 2012 09:35:32 -0700
Subject: [Freeswitch-users] FSClient and CELT codec version
In-Reply-To: <CAC-LwPMbLzi--HFOtE2Q3ziqB3o5q-sC+Y2ETK3vphB+8f2KMQ@mail.gmail.com>
References: <CAC-LwPMbLzi--HFOtE2Q3ziqB3o5q-sC+Y2ETK3vphB+8f2KMQ@mail.gmail.com>
Message-ID: <CALn233qZ9cF9Fxczz5U8qNKcugbEqk4k5BLD9pKyxiy_+Rb5tA@mail.gmail.com>

Make sure you select the CELT codec under options->Sofia Settings,
you can actually connect to FSClient using fs_cli on port 8022 by
default let me know if you still have issues.  FSClient is built
against 1.2 currently of FS.

~Mitch

On Mon, Sep 10, 2012 at 8:52 AM, Vik Killa <vipkilla at gmail.com> wrote:
> I'm trying to connect to a FS conference with FSClient using the CELT
> codec, unfortunately, it is not working due to some codec error.
> Im wondering if the problem is the difference in CELT versions. My
> FreeSWITCH server is running celt-0.10.0.
> What is FSClient running?
>
> 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 vipkilla at gmail.com  Mon Sep 10 20:49:39 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 10 Sep 2012 12:49:39 -0400
Subject: [Freeswitch-users] FSClient and CELT codec version
In-Reply-To: <CALn233qZ9cF9Fxczz5U8qNKcugbEqk4k5BLD9pKyxiy_+Rb5tA@mail.gmail.com>
References: <CAC-LwPMbLzi--HFOtE2Q3ziqB3o5q-sC+Y2ETK3vphB+8f2KMQ@mail.gmail.com>
	<CALn233qZ9cF9Fxczz5U8qNKcugbEqk4k5BLD9pKyxiy_+Rb5tA@mail.gmail.com>
Message-ID: <CAC-LwPNL2NQQfhQkata4_BdBpiyhi5uL8PQDHAO93pT1NsNebg@mail.gmail.com>

I've made sure that FSClient has the CELT codec selected. I connected
to the fs_cli for FSClient, it shows:
2012-09-10 12:48:17.311129 [ERR] switch_core_io.c:973 Codec CELT
ultra-low delay decoder error!

Any suggestions?
Thanks.

On Mon, Sep 10, 2012 at 12:35 PM, Mitch Capper <mitch.capper at gmail.com> wrote:
> Make sure you select the CELT codec under options->Sofia Settings,
> you can actually connect to FSClient using fs_cli on port 8022 by
> default let me know if you still have issues.  FSClient is built
> against 1.2 currently of FS.
>
> ~Mitch


From cmrienzo at gmail.com  Mon Sep 10 21:53:56 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 10 Sep 2012 13:53:56 -0400
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <A90437D5515AF341A7942C6446201F3E07154E5D94@DFW1MBX20.mex07a.mlsrvr.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
	<CAD-qRkjgK9rGFmznjVxM-CiCFFgYKBSTwOR5RwHevBAfaaviQA@mail.gmail.com>
	<A90437D5515AF341A7942C6446201F3E07154E5D94@DFW1MBX20.mex07a.mlsrvr.com>
Message-ID: <CAD-qRkhsj0O08WcrVkUACW1LtL--dqZMbSyQ89LiuYX6SGyiMA@mail.gmail.com>

Since the things that FS does (transcoding, tone detection, etc) can be CPU
intensive, you might want multiple servers to spread the load.  So, you can
have a pool of identically-configured FS servers load balanced by something
like opensips, with the (typically lighter-weight) application logic
handled outside of that pool of servers.  FS here just provides resources
for the application to use.

In this type of setup, you can pick between an application that actively
controls the FS apps to execute (mod_event_socket, mod_erlang_event) or one
that feeds FS higher-level scripts to execute on demand (mod_httapi,
mod_xml_curl).

Chris


On Mon, Sep 10, 2012 at 10:32 AM, Brett Clark - Grasshopper <
bclark at grasshopper.com> wrote:

> Hey Chris,****
>
> ** **
>
> Could you explain what you mean by clustering FS?  I get the idea of
> clustering, in general, but I what do you mean in this context?****
>
> ** **
>
> Thanks!****
>
> Brett****
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Christopher
> Rienzo
> *Sent:* Friday, September 07, 2012 12:14 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] best FS interface for development****
>
> ** **
>
> For arbitrarily large and complex systems, I'd keep FreeSWITCH as dumb as
> possible so that you can easily cluster them.  So, modules that allow
> external control (mod_event_socket, mod_erlang_event, mod_httapi,
> mod_xml_curl) are all good choices.
>
> Event socket is pretty popular; Plivo and Adhearsion both use it.  The
> 2600hz guys use mod_erlang_event.  And I've heard of plenty of people that
> use mod_xml_curl.
>
> Chris
>
>
> ****
>
> On Fri, Sep 7, 2012 at 9:43 AM, Brett Clark - Grasshopper <
> bclark at grasshopper.com> wrote:****
>
> There seems to be at least 4 general ways for customizing FS:****
>
> - Use the event socket and build an app to interface with it****
>
> - Use mod lua, or similar, to develop in a particular language****
>
> - Develop your own ?module? to interface directly with the core****
>
> - Use the XML interface to implement IVR, PBX, and voicemail type
> functionality and all the rest.****
>
>  ****
>
> How prevalent is each approach in the community?  It seems like everyone
> is doing a little of both, but maybe someone has a better handle of what
> the actual ratios are?  ****
>
>  ****
>
> Which is best suited for an arbitrarily large and complex application?  I
> realize this isn?t a simple question?what I want is to understand the most
> evolved and featureful way to interface with FS?  I don?t want to adopt an
> approach which isn?t being actively maintained or is missing features.  As
> new stuff is added to FS, which approach will allow me to adopt those new
> features most easily?****
>
>  ****
>
> Thanks!
> Brett****
>
>  ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From josh at foshee.info  Mon Sep 10 22:37:49 2012
From: josh at foshee.info (Joshua Foshee)
Date: Mon, 10 Sep 2012 13:37:49 -0500
Subject: [Freeswitch-users] Call recording speed issue
In-Reply-To: <504C2F2B.7030502@gmail.com>
References: <CAA=kHzLbobvFF737zV8Nn+DYrSu9dFommDG=o5ONeoatpPkBQA@mail.gmail.com>
	<504C2F2B.7030502@gmail.com>
Message-ID: <CAA=kHzKv3+ORo7iDuuyF49gLV+gntjxtkzaZxoY4fnixJNrh+Q@mail.gmail.com>

Well the phones I am using are Linksys SPA942. The Sip trunking I am using
is Flowroute.

The problem only happens with calls from flowroute to me. Then both the
flowroute side and the SPA942 side are fast on the recording.

START
Flowroute ------> Freeswitch -----> SPA942

If the call goes from SPA942 to flowroute then it sounds just fine in the
recording.

START
SPA942 ----> Freeswitch ------> Flowroute.

I have done a git and I am on the most current release

FreeSWITCH Version 1.3.0+git~20120908T211235Z~36cee285b0 (1.3.0; git at
commit 36cee285b0 on Sat, 08 Sep 2012 21:12:35 Z)


On Sun, Sep 9, 2012 at 12:54 AM, Vbvbrj <vbvbrj at gmail.com> wrote:

> On 09.09.2012 05:40, Joshua Foshee wrote:
> > We have a issue with the call quality of the recorded call to be in a
> > faster speed then what the call takes place in. I have took a packet
> > capture on the server to replay the audio stream and it sounds just
> > fine. Anyone have the same problem with the speed of the audio recording
>
> This happens because of transcoding. If you look carefully, the bad
> audio will be only for certain users' sip phones. Look for phones
> connected to problematic extensions and change in phone settings the
> signaling standard from china to other. Also update FS to latest 1.3.0
> release.
>
> --
> 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
>
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From vbvbrj at gmail.com  Tue Sep 11 00:51:46 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 10 Sep 2012 23:51:46 +0300
Subject: [Freeswitch-users] Call recording speed issue
In-Reply-To: <CAA=kHzKv3+ORo7iDuuyF49gLV+gntjxtkzaZxoY4fnixJNrh+Q@mail.gmail.com>
References: <CAA=kHzLbobvFF737zV8Nn+DYrSu9dFommDG=o5ONeoatpPkBQA@mail.gmail.com>
	<504C2F2B.7030502@gmail.com>
	<CAA=kHzKv3+ORo7iDuuyF49gLV+gntjxtkzaZxoY4fnixJNrh+Q@mail.gmail.com>
Message-ID: <504E52E2.7050801@gmail.com>

On 10.09.2012 21:37, Joshua Foshee wrote:
> Well the phones I am using are Linksys SPA942. The Sip trunking I am
> using is Flowroute.
>
> The problem only happens with calls from flowroute to me. Then both the
> flowroute side and the SPA942 side are fast on the recording.

Please make some test calls and open a jira issue with sound recorded 
posted, with configs and logs when this problems occur.

In my situation I don't see this problem anymore.

-- 
Mimiko desu.


From cabildo at gmail.com  Tue Sep 11 02:06:49 2012
From: cabildo at gmail.com (Julio Saldivar)
Date: Mon, 10 Sep 2012 18:06:49 -0400
Subject: [Freeswitch-users] Problem with record and script
Message-ID: <CAO2g7M-T4_re075YkkFT34KGud3MBGVDQ2fi8J7j68n4HOz9yA@mail.gmail.com>

Hello, I am trying to record a conversation, and then pass it to a script, but
it generates out the following error when the caller ends the call:

09.10.2012 18:47:07.651845 [DEBUG] switch_core_session.c: 2149
sofia/internal/3119 @ 192.168.0.225 Channel is hungup and Application '
python' does not Have the zombie_exec flag.

When the call ends the call everything works ok

<extension name="DISA for FS">
    <condition field="destination_number" expression="^(3499)$">
        <action application="set_zombie_exec"/>
            <action application="set" data="RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
            <action application="set" data="RECORD_COPYRIGHT=(c) 1980
Factory Records, Inc."/>
            <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
            <action application="set" data="RECORD_ARTIST=Ian Curtis"/>
            <action application="set" data="RECORD_COMMENT=Love will tear
us apart"/>
            <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
%H:%M)}"/>
            <action application="set" data="RECORD_STEREO=true"/>
            <action application="record_session" data="/tmp/${uuid}.wav"/>
        <action application="read" data="2 15
'tone_stream://%(10000,0,350,440)' digits 30000 #"/>
        <action application="execute_extension" data="${digits}"/>
                <action application="bridge"
data="sofia/gateway/cisco/${digits}"/>
        <action application="set_zombie_exec"/>
        <action application="python" data="disa1"/>
    </condition>
</extension>

-- 
Si alguna vez
mi voz deja de escucharse
piensen que el bosque
hablar? por m?
con su lenguaje de ra?ces.
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From msc at freeswitch.org  Tue Sep 11 03:52:08 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Sep 2012 16:52:08 -0700
Subject: [Freeswitch-users] Problem with record and script
In-Reply-To: <CAO2g7M-T4_re075YkkFT34KGud3MBGVDQ2fi8J7j68n4HOz9yA@mail.gmail.com>
References: <CAO2g7M-T4_re075YkkFT34KGud3MBGVDQ2fi8J7j68n4HOz9yA@mail.gmail.com>
Message-ID: <CAKzWOxVWDHOhS296E4UsBuQ_AhR2jzUrmkYPhUqC2rQt9Hn4Ow@mail.gmail.com>

Nice graphic, btw. :)

What are you trying to do with this channel after the caller hangs up? If
this is some sort of post processing then you need to call python from an
api_hangup_hook <http://wiki.freeswitch.org/wiki/Variable_api_hangup_hook>.

-MC


On Mon, Sep 10, 2012 at 3:06 PM, Julio Saldivar <cabildo at gmail.com> wrote:

> Hello, I am trying to record a conversation, and then pass it to a script, but
> it generates out the following error when the caller ends the call:
>
> 09.10.2012 18:47:07.651845 [DEBUG] switch_core_session.c: 2149
> sofia/internal/3119 @ 192.168.0.225 Channel is hungup and Application '
> python' does not Have the zombie_exec flag.
>
> When the call ends the call everything works ok
>
> <extension name="DISA for FS">
>     <condition field="destination_number" expression="^(3499)$">
>         <action application="set_zombie_exec"/>
>             <action application="set" data="RECORD_TITLE=Recording
> ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
>             <action application="set" data="RECORD_COPYRIGHT=(c) 1980
> Factory Records, Inc."/>
>             <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>             <action application="set" data="RECORD_ARTIST=Ian Curtis"/>
>             <action application="set" data="RECORD_COMMENT=Love will tear
> us apart"/>
>             <action application="set"
> data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
>             <action application="set" data="RECORD_STEREO=true"/>
>             <action application="record_session" data="/tmp/${uuid}.wav"/>
>         <action application="read" data="2 15
> 'tone_stream://%(10000,0,350,440)' digits 30000 #"/>
>         <action application="execute_extension" data="${digits}"/>
>                 <action application="bridge"
> data="sofia/gateway/cisco/${digits}"/>
>         <action application="set_zombie_exec"/>
>         <action application="python" data="disa1"/>
>     </condition>
> </extension>
>
> --
> Si alguna vez
> mi voz deja de escucharse
> piensen que el bosque
> hablar? por m?
> con su lenguaje de ra?ces.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 royj at yandex.ru  Tue Sep 11 12:48:16 2012
From: royj at yandex.ru (royj)
Date: Tue, 11 Sep 2012 12:48:16 +0400
Subject: [Freeswitch-users] t38 issue
Message-ID: <20120911124816.82cafdb7fa10697b81903e7b@yandex.ru>


Hi all

There is some strange issue with t38-passthru (FreeSWITCH Version 1.2.0-rc2).
Task - t38-passthru
spa112(ip 1.1.1.1) <--(profile local)--> FreeSWITCH(ip 2.2.2.2)<--(profile extrernal)--> Mediant 2000(ip 3.3.3.3); fax from spa112 to Mediant 2000.
In both profiles:
<param name="t38-passthru" value="true"/>
<param name="inbound-proxy-media" value="true"/>
<param name="inbound-late-negotiation" value="true"/>

Here is the sip-trace of the call with fax - http://pastebin.freeswitch.org/19866

It is seen that t38 re-INVITE from spa112 (profile local) passed on and back with "200 OK", but t38 from Mediant 2000 (profile external) is not (in response to re-INVITE t38 (SDP, m=image) - "200 OK" with SDP m=audio). After agreement the SDP, the data (v21-preamble) is not passed by freeswitch between spa112 and Mediant 2000, and one of the party ends conversation with message "no-signal" and then "by" .

Wiki says ( http://wiki.freeswitch.org/wiki/Proxy_Media#Why_you_almost_certainly_don.27t_want_to_use_it ) that module mod_spandsp is required (if I understand correctly), but "load mod_spandsp" does not change anything.

I would appreciate your help

-- 
Regards
royj


From tonybecq at yahoo.fr  Tue Sep 11 14:13:03 2012
From: tonybecq at yahoo.fr (Tony BECQ)
Date: Tue, 11 Sep 2012 11:13:03 +0100 (BST)
Subject: [Freeswitch-users]  simulating a modem transaction...
Message-ID: <1347358383.58518.YahooMailNeo@web171202.mail.ir2.yahoo.com>

Hi

I've looked for a good tutorial on spanDSP but haven't found one. In my former message (link) I've talked about the way the modems have to connect each other and exchange some informations.

I'm still looking for a good tutorial about spanDSP,
I need to know if what I've described in my former message about de connection between the two modems is a standard one or not and if it is not, how I can use spandsp library to program a correct one for my needs.

Thanks in advance...

Tony
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From tonybecq at yahoo.fr  Tue Sep 11 14:21:05 2012
From: tonybecq at yahoo.fr (Tony BECQ)
Date: Tue, 11 Sep 2012 11:21:05 +0100 (BST)
Subject: [Freeswitch-users]  simulating a modem transaction...
Message-ID: <1347358865.61451.YahooMailNeo@web171205.mail.ir2.yahoo.com>


Hi

I've looked for a good tutorial on spanDSP but haven't found 
one. In my former message I've talked about the way the modems have to 
connect each other and exchange some informations.

I'm still looking for a good tutorial about spanDSP,
I
 need to know if what I've described in my former message about de 
connection between the two modems is a standard processus or not and if it is 
not, how I can use spandsp library to program a correct one for my 
needs.

Thanks in advance...

Tony


>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
>
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From mike at jerris.com  Tue Sep 11 17:29:41 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Sep 2012 09:29:41 -0400
Subject: [Freeswitch-users] mod_shout / telecast
In-Reply-To: <D1925DECD316894DAADB84076C1B7D827E2951B3@exchange.besharp.at>
References: <D1925DECD316894DAADB84076C1B7D827E2950A2@exchange.besharp.at>
	<D1925DECD316894DAADB84076C1B7D827E295143@exchange.besharp.at>
	<A27FE7A21D714632A76C01A041E66A51@gmail.com>
	<D1925DECD316894DAADB84076C1B7D827E2951B3@exchange.besharp.at>
Message-ID: <D657EA22-CD08-47B7-AEF8-D5499F7BF9AA@jerris.com>

It's on my short list to apply this.  Thanks for the feedback

Mike

On Sep 7, 2012, at 8:29 AM, Gerald Weber <gerald.weber at besharp.at> wrote:

> I applied the patch before i wrote my last mail, and it works fine.
> I commented and voted on jira now J
>  
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Seven Du
> Gesendet: Freitag, 07. September 2012 13:47
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_shout / telecast
>  
> - apply the patch and see if it helps
>  
> - comment on the jira to get more attention
>  On Friday, September 7, 2012 at 7:40 PM, Gerald Weber wrote:
> I answer that myself:
> There is a jira for this, http://jira.freeswitch.org/browse/FS-3960
> Seems like /webapi produces some other output than /api
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gerald Weber
> Gesendet: Freitag, 07. September 2012 12:21
> An: freeswitch-users at lists.freeswitch.org
> Betreff: [Freeswitch-users] mod_shout / telecast
>  
> Hi,
>  
> I?m trying to embed the mp3 link from http://x.x.x.x:8080/webapi/telecast/mp3/72d6e296-f8d4-11e1-b7b6-f1154fdbf73a/2001.mp3
> (from http://x.x.x.x:8080/webapi/telecast/index ) into an embedded player object in a html page.
> (i?m on latest git)
>  
> The problem: i?m getting garbage as output from the above link, regardless which browser i use.
> Even winamp doesnt recognize the stream.
>  
> Is mod_shout / telecast currently broken or am i doing something wrong ?
> 
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From dgarcia at anew.com.ve  Tue Sep 11 17:31:47 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Tue, 11 Sep 2012 09:01:47 -0430
Subject: [Freeswitch-users] simultaneous calls
In-Reply-To: <20120910031044.GE15903@ztillma.com>
References: <20120910031044.GE15903@ztillma.com>
Message-ID: <504F3D43.6040805@anew.com.ve>

Hi,

Are you triggering your originate from console? Perhaps you should 
change a little bit how you do it. I think if  you want really to 
originate a call simultaneously to two separate phone, you should use 
some sort of programming tools (php, lua, java, javascript, c, c++, etc) 
and even use other Freeswitch tools (esl, mod_lua, mod_httapi, etc).

I will try to explain a little more by an example. You could use php, 
php in linux have a lib to manage multiple task simultaneously; so with 
php you could fire several originate against FS at the same time.

On 9/9/2012 10:40 PM, Zaid Tillma wrote:
> My goal is to have two phone announce a recording (in a wav file). I have accomplished goal in the following manner:
>
> I have an extension configured like this:
>
> <extension name="wavs">
>      <condition field="destination_number" expression="^2009$">
>        <action application="playback" data="/path/to/your.wav"/>
>      </condition>
>    </extension>
>
> When you call, you get an audio playback.
>
> I am trying to originate a call simultaneously to two separate phone (i.e want to two phone to play audio) at the same time, and  I am doing it like this (and it works):
>
> /usr/local/freeswitch/bin/fs_cli -x 'originate {ignore_early_media=true,sip_auto_answer=true}user/1000  2009'
> /usr/local/freeswitch/bin/fs_cli -x 'originate {ignore_early_media=true,sip_auto_answer=true}user/1001  2009'
>
> There's got to be a better way to do this way without having two seperate originates.. I did try the Enterprise originate, but only one phone answered... Not sure if there was a mistake in my Enterprise originate implementation.
>
> Please let me know if there is a more elegant way to do this..
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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: 2012.0.2197 / Virus Database: 2437/5254 - Release Date: 09/07/12
>
>


-- 
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 erik.dekkers at certhon.com  Tue Sep 11 17:36:43 2012
From: erik.dekkers at certhon.com (Erik Dekkers)
Date: Tue, 11 Sep 2012 15:36:43 +0200
Subject: [Freeswitch-users] XMPP Presence
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE3951668588@FS1.wvds.local>

Hi Guys,

I?m not sure if this has been on the mailinglist before, but i couldn?t find it on the search.

My situation is like this:

Running a Freeswitch server as our PBX, connecting Aastra phones to it.
Right now I would like to use an Attendant/Operator console application and downloaded Voice Operator Panel (wich looks really good)
The presence part of the Voice Operator Panel is working with XMPP.

Is it possible (with or without a 3rd party XMPP server) to send presence events so the Voice Operator Panel can track the presence of users?
Or just another attendant/operator application :)

Thnx and best regards,

Erik Dekkers

[cid:imagec0e560.PNG at bcafa31c.47b283a9]

ABC Westland 555        Tel:    +31 174 22 50 80
P.O. Box 90     Fax:    +31 174 22 50 81
        Mob:    +31 624 423 009
2685 ZH Poeldijk        erik.dekkers at certhon.com<mailto:erik.dekkers at certhon.com>
The Netherlands www.certhon.com<http://www.certhon.com>

DISCLAIMER
All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS.
Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing.

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From ssoni at lifesize.com  Tue Sep 11 14:12:29 2012
From: ssoni at lifesize.com (Sanjay Soni)
Date: Tue, 11 Sep 2012 05:12:29 -0500
Subject: [Freeswitch-users] Help on freeswitch : gtalk-sip conversion
Message-ID: <F3C02C5E8650B74FA0C199FA1A01D26E16482CD4BB@ausmsex00.austin.kmvtechnologies.com>

Hi,
I am trying to get a call working from Gtalk client (Running on my laptop) to the SIP device(Registered as user name 1000). When gtalk client make a call, call is accepted by SIP device but the media is not flowing between Freeswitch and gtalk client.
Please note that all the three (Freeswitch machine, gtalk client and SIP device) are in same private LAN inside the firewall. I see that freeswitch is rejecting the local candidate sent by gtalk client and selecting the stun candidate. Why is so ? How can I stop it from doing this. If it could select local candidate I feel call could establish.

Thanks
Sanjay


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From msc at freeswitch.org  Tue Sep 11 19:35:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Sep 2012 08:35:27 -0700
Subject: [Freeswitch-users] t38 issue
In-Reply-To: <20120911124816.82cafdb7fa10697b81903e7b@yandex.ru>
References: <20120911124816.82cafdb7fa10697b81903e7b@yandex.ru>
Message-ID: <CAKzWOxVzz3-nmYW-uOGaeKE25P53sNr8MzAUBn_2_sAakVtubQ@mail.gmail.com>

I think the guys have made some improvements since 1.2.0-rc2. Any chance
you can get at least up to 1.2.1-stable and retest?
-MC

On Tue, Sep 11, 2012 at 1:48 AM, royj <royj at yandex.ru> wrote:

>
> Hi all
>
> There is some strange issue with t38-passthru (FreeSWITCH Version
> 1.2.0-rc2).
> Task - t38-passthru
> spa112(ip 1.1.1.1) <--(profile local)--> FreeSWITCH(ip 2.2.2.2)<--(profile
> extrernal)--> Mediant 2000(ip 3.3.3.3); fax from spa112 to Mediant 2000.
> In both profiles:
> <param name="t38-passthru" value="true"/>
> <param name="inbound-proxy-media" value="true"/>
> <param name="inbound-late-negotiation" value="true"/>
>
> Here is the sip-trace of the call with fax -
> http://pastebin.freeswitch.org/19866
>
> It is seen that t38 re-INVITE from spa112 (profile local) passed on and
> back with "200 OK", but t38 from Mediant 2000 (profile external) is not (in
> response to re-INVITE t38 (SDP, m=image) - "200 OK" with SDP m=audio).
> After agreement the SDP, the data (v21-preamble) is not passed by
> freeswitch between spa112 and Mediant 2000, and one of the party ends
> conversation with message "no-signal" and then "by" .
>
> Wiki says (
> http://wiki.freeswitch.org/wiki/Proxy_Media#Why_you_almost_certainly_don.27t_want_to_use_it) that module mod_spandsp is required (if I understand correctly), but
> "load mod_spandsp" does not change anything.
>
> I would appreciate your help
>
> --
> Regards
> royj
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Sep 11 20:00:43 2012
From: all.eforums at gmail.com (A E G)
Date: Tue, 11 Sep 2012 12:00:43 -0400
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <CAD-qRkhsj0O08WcrVkUACW1LtL--dqZMbSyQ89LiuYX6SGyiMA@mail.gmail.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
	<CAD-qRkjgK9rGFmznjVxM-CiCFFgYKBSTwOR5RwHevBAfaaviQA@mail.gmail.com>
	<A90437D5515AF341A7942C6446201F3E07154E5D94@DFW1MBX20.mex07a.mlsrvr.com>
	<CAD-qRkhsj0O08WcrVkUACW1LtL--dqZMbSyQ89LiuYX6SGyiMA@mail.gmail.com>
Message-ID: <CADzY+JjCVxv9mdffMBzQv855OO_H2pjW_XovHDVzb_CCVY0+Xg@mail.gmail.com>

I like this discussion. Unfortunately I'm too inexperienced to fully
understand why one method/interface would be used over the other. Is this a
matter of "inbound" vs "outbound" interfaces? Or is this a matter of taking
over the control of each call and making all decisions dynamically on
behalf of FS as opposed to say doing a relatively "static" configuration of
FS by sending it an XML on the fly to process as opposed to api commands
over socket as you would in the case of mod_event_socket?

Are they all variations for the same purpose and so the decision to use one
or the other depends on your application and how it's able to interface
with FS? or is there also another level of decision where one is clearly
better/powerful/faster/efficient than the other, and that could infact
drive the decision for your application development?

Would it fair to compare the functionality and interface offered by
mod_httapi to that of something like Plivo, in terms of abstraction of the
most common functions required from FS by simply interacting with commands
like say, speak, play, record etc? and that this is at a higher level than
mod_xml_curl?

Thanks (and apologies for the ignorance)

On Mon, Sep 10, 2012 at 1:53 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> Since the things that FS does (transcoding, tone detection, etc) can be
> CPU intensive, you might want multiple servers to spread the load.  So, you
> can have a pool of identically-configured FS servers load balanced by
> something like opensips, with the (typically lighter-weight) application
> logic handled outside of that pool of servers.  FS here just provides
> resources for the application to use.
>
> In this type of setup, you can pick between an application that actively
> controls the FS apps to execute (mod_event_socket, mod_erlang_event) or one
> that feeds FS higher-level scripts to execute on demand (mod_httapi,
> mod_xml_curl).
>
> Chris
>
> On Mon, Sep 10, 2012 at 10:32 AM, Brett Clark - Grasshopper <
> bclark at grasshopper.com> wrote:
>
>> Hey Chris,****
>>
>> ** **
>>
>> Could you explain what you mean by clustering FS?  I get the idea of
>> clustering, in general, but I what do you mean in this context?****
>>
>> ** **
>>
>> Thanks!****
>>
>> Brett****
>>
>> ** **
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Christopher
>> Rienzo
>> *Sent:* Friday, September 07, 2012 12:14 PM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] best FS interface for development****
>>
>> ** **
>>
>> For arbitrarily large and complex systems, I'd keep FreeSWITCH as dumb as
>> possible so that you can easily cluster them.  So, modules that allow
>> external control (mod_event_socket, mod_erlang_event, mod_httapi,
>> mod_xml_curl) are all good choices.
>>
>> Event socket is pretty popular; Plivo and Adhearsion both use it.  The
>> 2600hz guys use mod_erlang_event.  And I've heard of plenty of people that
>> use mod_xml_curl.
>>
>> Chris
>>
>>
>> ****
>>
>> On Fri, Sep 7, 2012 at 9:43 AM, Brett Clark - Grasshopper <
>> bclark at grasshopper.com> wrote:****
>>
>> There seems to be at least 4 general ways for customizing FS:****
>>
>> - Use the event socket and build an app to interface with it****
>>
>> - Use mod lua, or similar, to develop in a particular language****
>>
>> - Develop your own ?module? to interface directly with the core****
>>
>> - Use the XML interface to implement IVR, PBX, and voicemail type
>> functionality and all the rest.****
>>
>>  ****
>>
>> How prevalent is each approach in the community?  It seems like everyone
>> is doing a little of both, but maybe someone has a better handle of what
>> the actual ratios are?  ****
>>
>>  ****
>>
>> Which is best suited for an arbitrarily large and complex application?  I
>> realize this isn?t a simple question?what I want is to understand the most
>> evolved and featureful way to interface with FS?  I don?t want to adopt an
>> approach which isn?t being actively maintained or is missing features.  As
>> new stuff is added to FS, which approach will allow me to adopt those new
>> features most easily?****
>>
>>  ****
>>
>> Thanks!
>> Brett****
>>
>>  ****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Sep 11 20:02:30 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 11 Sep 2012 09:02:30 -0700
Subject: [Freeswitch-users] TDM to SIP Gateway
Message-ID: <11FEA525-0BBA-42CB-BD91-3ADB1FC3A566@5ninesolutions.com>

Hello,
I have a customer with a Comdial PBX that we would like to connect using SIP trunks.  The Comdial is currently connected via ISDN PRI.  Before the numbers are ported, the existing PRI will need to be active for inbound calls.  Can I use a Freeswitch server with a 2 (one port to PBX, one to PSTN) T1 Sangoma card to do this?  Would I need hardware echo cancelation since the cable run would only be about 10 feet?  Are there any directional (i.e. FXO/FXS in the analog world) or timing considerations?

Forgive my ignorance, my background is primarily on the SIP side of things.

Thanks for any help,
Spencer



From curriegrad2004 at gmail.com  Tue Sep 11 20:09:09 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 11 Sep 2012 09:09:09 -0700
Subject: [Freeswitch-users] TDM to SIP Gateway
In-Reply-To: <11FEA525-0BBA-42CB-BD91-3ADB1FC3A566@5ninesolutions.com>
References: <11FEA525-0BBA-42CB-BD91-3ADB1FC3A566@5ninesolutions.com>
Message-ID: <CAMWiMy986onSBTGhdSKqVMrGzKkZUhbrrd+4c7UUNWBs3dQqVQ@mail.gmail.com>

If it's a TDM link, there aren't really any special considerations
other than setting the Line Build Out to < 150ft (I forgot the
specific details here, but choose the appropriate levels). In the old
TDM world, there is no FXO/FXS considerations but there is a huge
thing about clocking.

For the telco side of your PRI, you'll have to set your end as the
slave and the PRI that's going towards your PBX has to be set as
master and having your legacy PBX being set as a slave.

Btw, North American PRI's are really T1's in disguise, so yes you can
go ahead and use your existing T1 cards if you have one. ;)

On Tue, Sep 11, 2012 at 9:02 AM, Spencer Thomason
<spencer at 5ninesolutions.com> wrote:
> Hello,
> I have a customer with a Comdial PBX that we would like to connect using SIP trunks.  The Comdial is currently connected via ISDN PRI.  Before the numbers are ported, the existing PRI will need to be active for inbound calls.  Can I use a Freeswitch server with a 2 (one port to PBX, one to PSTN) T1 Sangoma card to do this?  Would I need hardware echo cancelation since the cable run would only be about 10 feet?  Are there any directional (i.e. FXO/FXS in the analog world) or timing considerations?
>
> Forgive my ignorance, my background is primarily on the SIP side of things.
>
> Thanks for any help,
> Spencer
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From dujinfang at gmail.com  Tue Sep 11 20:26:45 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 12 Sep 2012 00:26:45 +0800
Subject: [Freeswitch-users] xml tags
Message-ID: <39810AE90DFB4F3399336768F94B9CB9@gmail.com>

Hi, 

in conference xml_list some tags has "_" but some use "-"

"_" works well but "-" causing problems e.g. when convert to json it's ok to use conference.last_talking but to get the input-volume value it has to be conference["input-volume"], and in some other cased there's no such work around and I have to go through all tags and replace "-"s to "_"s.

I understand "-" in xml is totally valid but would it be better to keep it consistent ? 

<last_talking>0</last_talking> <energy>300</energy> <volume_in>0</volume_in> <volume_out>0</volume_out> <output-volume>0</output-volume> <input-volume>0</input-volume> <auto-adjusted-input-volume>0</auto-adjusted-input-volume>


Thanks.

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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From Rob.Moore at Aeriandi.com  Tue Sep 11 22:11:03 2012
From: Rob.Moore at Aeriandi.com (Rob Moore)
Date: Tue, 11 Sep 2012 18:11:03 +0000
Subject: [Freeswitch-users] Altering From Header in SIP Invite
Message-ID: <49C5FCA19A8A114493EBAACA42FE5899105C87C2@1AERDCEXCHMBX1.AER.AERCO.local>

Hi All,

I'm having a little trouble with 'presentation numbers' with a new provider I'm in IOT with this week.

I'm trying to recreate the following Invite as the calls pass through our dialplan to this provider but there are issues with trying to get a different CLI into the From and P-Asserted-Identity headers.

When presenting a Freephone number (for example) we need to still present the regular number that would be used by this extension in the P-Asserted-Identity whilst passing the number we wish to have presented in the From header.

Currently we are not using Gateways so we cannot resort to using  <param name="caller-id-in-from" value="true"/> (although I expect this won't do what we need in this case) so I've looked at altering the channel variables  sip_from_user,sip_full_from and sip_full_uri using set and export dial plan apps but none of these seem to have any effect so I guess these variables must be read only.

I'm sure this must be simple, but can't for the life of me work out what I need to do.

Below is an extract from an example header from the provider I am trying to recreate, I've also added a copy of the Dialplan extension I am using to test.
If someone can tell me what I'm getting wrong I would really appreciate it.

Thanks

Rob


INVITE sip:+445600005262 at primarysip.barfoo.com;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: +445600000262 <sip:+445600005262 at primarysip.barfoo.com;user=phone>
From: <sip:+448000655262 at uk.foo.bar.2.net;user=phone>;tag=12544
P-Asserted-Identity: <sip:+445600655262 at uk.sdin.bt.net>
Call-ID: 1347372978-13100 at mgc-uk-998.n2<mailto:1347372978-13100 at mgc-uk-998.n2>
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE


DIAL Plan: (attempting +445600655262 in P Asserted-Id and +448000655262 in from) have commented out some things that I have tried.)
If you are worried about [sip_h_X-Gateway=4003:4] this is added to help our SBC forward calls to our different providers.

   <extension name="TC 25">
     <condition field="destination_number" expression="^10125$">
                                <action application="set" data="effective_caller_id_name=_undef_"/>
                                <action application="set" data="effective_caller_id_number=+445600005262 "/>
                                <action application="set" data="dtmf_type=rfc2833"/>
                                <action application="set" data="sip_cid_type=pid"/>
                                <!--<action application="export" data=" sip_from_user =+448000655262"/> -->
                                <!-- <action application="export" data="sip_h_from=<sip:+448000655262 at primarysip.barfoo.com>;tag=${sip_from_tag} "/> -->
                                <!--  <action application="export" data="sip_h_P-Asserted-Identity= +445600005262 <sip:+445600655262 at 172.17.7.4> "/> -->
                                <action application="info"/>
                  <action application="bridge" data="[sip_h_X-Gateway=4003:4]sofia/internal/+442920001199 at pstn.tel" />
     </condition>
   </extension>


Rob Moore
Telephony Systems Infrastructure Manager
Aeriandi
Aeriandi Ltd, Prama House, Banbury Road, Oxford, OX27HT
E: rob.moore at aeriandi.com<mailto:rob.moore at aeriandi.com>
W: www.aeriandi.com<http://www.aeriandi.com/>
M: +44 (0)7766 838040
T: +44 (0) 845 108 0308

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From msc at freeswitch.org  Tue Sep 11 22:21:54 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Sep 2012 11:21:54 -0700
Subject: [Freeswitch-users] best FS interface for development
In-Reply-To: <CADzY+JjCVxv9mdffMBzQv855OO_H2pjW_XovHDVzb_CCVY0+Xg@mail.gmail.com>
References: <A90437D5515AF341A7942C6446201F3E07154E5A1C@DFW1MBX20.mex07a.mlsrvr.com>
	<CAD-qRkjgK9rGFmznjVxM-CiCFFgYKBSTwOR5RwHevBAfaaviQA@mail.gmail.com>
	<A90437D5515AF341A7942C6446201F3E07154E5D94@DFW1MBX20.mex07a.mlsrvr.com>
	<CAD-qRkhsj0O08WcrVkUACW1LtL--dqZMbSyQ89LiuYX6SGyiMA@mail.gmail.com>
	<CADzY+JjCVxv9mdffMBzQv855OO_H2pjW_XovHDVzb_CCVY0+Xg@mail.gmail.com>
Message-ID: <CAKzWOxVy8gSsyBasPxvPC5BJGNO=uxjLZ139hhVV4uySzNOERg@mail.gmail.com>

These are all excellent questions. In fact, this is one of the reasons why
Dave Kompel is working on the "10,000 foot view" document that we'll be
publishing, hopefully soon. (Listen to last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_05>for more
information.) Hopefully that will help you to mentally separate
things that may be unnecessarily jumbled together in your mind.

Here's an incomplete but hopefully helpful overview.  You can "control"
FreeSWITCH functionality in a few basic ways:
* Controlling call routing (where calls go, i.e. connecting party A to
party B)
* Controlling call actions (what calls actually do, i.e. IVRs)
* Controlling non-dialplan configuration, like users, conferences,
SIP/sofia, etc.

Then you have tools for accomplishing the control above.
* Static XML for everything
* Static XML for some things and dynamic for others (like dialplan,
directory, etc.)
** dialplan uses mod_xml_curl
** dialplan uses mod_httapi
** dialplan uses a scripting language to generate dialplan config
* Dynamic config of everything using mod_xml_curl
* Dynamic config with static "fall back" in case of mod_xml_curl lookup
failure

As a subset of "configuration" you have call control. Technically it can be
part of the FreeSWITCH "configuration" but it doesn't have to be. When I
say "call control" I don't mean "call routing." In this discussion "call
routing" is just getting a call from A to B and "call control" means
interacting with a call, usually with IVR-like functionality. Here are some
options:

Static XML for everything, i.e. XML dialplan for "call routing" and for
"call control"
Static XML for call routing, Lua/Perl/Python dialplan script for "call
control"
Static XML for call routing, socket-based ESL script for "call control"
Static XML for call routing, mod_httapi for "call control"
(You can also do dynamic XML in these cases :)

I hope the above information is useful in breaking things down a bit. So,
when someone says, "How can I control FreeSWITCH?" you understand why we
ask all of these follow up questions.

When someone asks, "What's the best way to do call control?" the correct
answer has been the same since the dawn of time: "It depends." There are
too many variables to give a simple answer. However, here are some rules of
thumb that might help you narrow it down:

* Do simple stuff with XML dialplan (static, dynamic - that's up to you)
* Do more complex logic with a dialplan script (I prefer Lua)
* If you need absolute control of a call leg, and especially if you plan on
bridging that call leg to other calls legs that you wish to control then
avoid dialplan scripts! I prefer ESL in these cases.

Naturally there are other things to consider. For example, would mod_httapi
work for you? Its focus is more narrow than ESL but with that narrower
focus comes less complexity. Maybe that's what you want or need. In any
case, I hope this is good food for thought. I'm sure the substance of this
conversation will make it into the big picture document that we are trying
to create.

-MC


On Tue, Sep 11, 2012 at 9:00 AM, A E G <all.eforums at gmail.com> wrote:

> I like this discussion. Unfortunately I'm too inexperienced to fully
> understand why one method/interface would be used over the other. Is this a
> matter of "inbound" vs "outbound" interfaces? Or is this a matter of taking
> over the control of each call and making all decisions dynamically on
> behalf of FS as opposed to say doing a relatively "static" configuration of
> FS by sending it an XML on the fly to process as opposed to api commands
> over socket as you would in the case of mod_event_socket?
>
> Are they all variations for the same purpose and so the decision to use
> one or the other depends on your application and how it's able to interface
> with FS? or is there also another level of decision where one is clearly
> better/powerful/faster/efficient than the other, and that could infact
> drive the decision for your application development?
>
> Would it fair to compare the functionality and interface offered by
> mod_httapi to that of something like Plivo, in terms of abstraction of the
> most common functions required from FS by simply interacting with commands
> like say, speak, play, record etc? and that this is at a higher level than
> mod_xml_curl?
>
> Thanks (and apologies for the ignorance)
>
> On Mon, Sep 10, 2012 at 1:53 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:
>
>> Since the things that FS does (transcoding, tone detection, etc) can be
>> CPU intensive, you might want multiple servers to spread the load.  So, you
>> can have a pool of identically-configured FS servers load balanced by
>> something like opensips, with the (typically lighter-weight) application
>> logic handled outside of that pool of servers.  FS here just provides
>> resources for the application to use.
>>
>> In this type of setup, you can pick between an application that actively
>> controls the FS apps to execute (mod_event_socket, mod_erlang_event) or one
>> that feeds FS higher-level scripts to execute on demand (mod_httapi,
>> mod_xml_curl).
>>
>> Chris
>>
>> On Mon, Sep 10, 2012 at 10:32 AM, Brett Clark - Grasshopper <
>> bclark at grasshopper.com> wrote:
>>
>>> Hey Chris,****
>>>
>>> ** **
>>>
>>> Could you explain what you mean by clustering FS?  I get the idea of
>>> clustering, in general, but I what do you mean in this context?****
>>>
>>> ** **
>>>
>>> Thanks!****
>>>
>>> Brett****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Christopher
>>> Rienzo
>>> *Sent:* Friday, September 07, 2012 12:14 PM
>>>
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] best FS interface for development****
>>>
>>> ** **
>>>
>>> For arbitrarily large and complex systems, I'd keep FreeSWITCH as dumb
>>> as possible so that you can easily cluster them.  So, modules that allow
>>> external control (mod_event_socket, mod_erlang_event, mod_httapi,
>>> mod_xml_curl) are all good choices.
>>>
>>> Event socket is pretty popular; Plivo and Adhearsion both use it.  The
>>> 2600hz guys use mod_erlang_event.  And I've heard of plenty of people that
>>> use mod_xml_curl.
>>>
>>> Chris
>>>
>>>
>>> ****
>>>
>>> On Fri, Sep 7, 2012 at 9:43 AM, Brett Clark - Grasshopper <
>>> bclark at grasshopper.com> wrote:****
>>>
>>> There seems to be at least 4 general ways for customizing FS:****
>>>
>>> - Use the event socket and build an app to interface with it****
>>>
>>> - Use mod lua, or similar, to develop in a particular language****
>>>
>>> - Develop your own ?module? to interface directly with the core****
>>>
>>> - Use the XML interface to implement IVR, PBX, and voicemail type
>>> functionality and all the rest.****
>>>
>>>  ****
>>>
>>> How prevalent is each approach in the community?  It seems like everyone
>>> is doing a little of both, but maybe someone has a better handle of what
>>> the actual ratios are?  ****
>>>
>>>  ****
>>>
>>> Which is best suited for an arbitrarily large and complex application?
>>> I realize this isn?t a simple question?what I want is to understand the
>>> most evolved and featureful way to interface with FS?  I don?t want to
>>> adopt an approach which isn?t being actively maintained or is missing
>>> features.  As new stuff is added to FS, which approach will allow me to
>>> adopt those new features most easily?****
>>>
>>>  ****
>>>
>>> Thanks!
>>> Brett****
>>>
>>>  ****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Sep 11 22:36:01 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 11 Sep 2012 13:36:01 -0500
Subject: [Freeswitch-users] t38 issue
In-Reply-To: <CAKzWOxVzz3-nmYW-uOGaeKE25P53sNr8MzAUBn_2_sAakVtubQ@mail.gmail.com>
Message-ID: <CC74EEC1.4C1B4%krice@freeswitch.org>

There have been updates to t.38 since 1.2.0-rc2...  You should either be on
head or v1.2.stable branch




On 9/11/12 10:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:

> I think the guys have made some improvements since 1.2.0-rc2. Any chance you
> can get at least up to 1.2.1-stable and retest?
> -MC
> 
> On Tue, Sep 11, 2012 at 1:48 AM, royj <royj at yandex.ru> wrote:
>> 
>> Hi all
>> 
>> There is some strange issue with t38-passthru (FreeSWITCH Version 1.2.0-rc2).
>> Task - t38-passthru
>> spa112(ip 1.1.1.1) <--(profile local)--> FreeSWITCH(ip 2.2.2.2)<--(profile
>> extrernal)--> Mediant 2000(ip 3.3.3.3); fax from spa112 to Mediant 2000.
>> In both profiles:
>> <param name="t38-passthru" value="true"/>
>> <param name="inbound-proxy-media" value="true"/>
>> <param name="inbound-late-negotiation" value="true"/>
>> 
>> Here is the sip-trace of the call with fax -
>> http://pastebin.freeswitch.org/19866
>> 
>> It is seen that t38 re-INVITE from spa112 (profile local) passed on and back
>> with "200 OK", but t38 from Mediant 2000 (profile external) is not (in
>> response to re-INVITE t38 (SDP, m=image) - "200 OK" with SDP m=audio). After
>> agreement the SDP, the data (v21-preamble) is not passed by freeswitch
>> between spa112 and Mediant 2000, and one of the party ends conversation with
>> message "no-signal" and then "by" .
>> 
>> Wiki says ( 
>> http://wiki.freeswitch.org/wiki/Proxy_Media#Why_you_almost_certainly_don.27t_
>> want_to_use_it ) that module mod_spandsp is required (if I understand
>> correctly), but "load mod_spandsp" does not change anything.
>> 
>> I would appreciate your help
>> 
>> --
>> Regards
>> royj
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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.burlingame at me.com  Tue Sep 11 22:39:31 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Tue, 11 Sep 2012 11:39:31 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <C640A2A2-7548-448F-8DCF-C0918289B09B@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
Message-ID: <9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>

Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax

Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared 
 
On Sep 8, 2012, at 9:36 AM, Mike Burlingame <mike.burlingame at me.com> wrote:

> Ok got another crash oddly enough it's kinda the same as yesterday's 
> 
> Yesterday the last message I see in the log file is a 401 Unauthorized before the log file stopped recording data and fs_cli exited to prompt today same thing happened but this time it looks like it did a core dump.
> 
> Opening JIRA case now
> 
> 
> On Sep 7, 2012, at 10:23 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
> 
>> I figured I would update to the latest git due to in the past I have seen "make sure your running the latest git" the odd thing is no core dump was created on the last crash.
>> 
>> I will post the backtrace if a core dump is generated after the next crash
>> 
>> 
>> On Sep 7, 2012, at 10:16 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> FYI: You need to get the backtrace before you update or the core file becomes useless.
>>> It dumps core by default so you should have one but if you updated, its tool late unless you can checkout the exact previous rev.
>>> 
>>> 
>>> 
>>> On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> after putting the server in production and  loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.
>>> 
>>> However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.
>>> 
>>> 
>>> 
>>> On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> 
>>> > After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
>>> >
>>> > Thanks
>>> >
>>> > On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> >
>>> >> Looks much much better Thank you -- Now to conduct more testing
>>> >>
>>> >> 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-Leg:5060, with session description
>>> >> 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying
>>> >> 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>> >> 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>> >> 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >> 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing
>>> >> 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >> 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >> 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description
>>> >> 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> >> 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> >> 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> >> 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK
>>> >> 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> >> 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
>>> >>
>>> >>
>>> >>
>>> >> On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> >>
>>> >>> ok,
>>> >>>
>>> >>> update one more time, if it still does not work just go right to jira
>>> >>> with the latest (not before today's changes)
>>> >>>
>>> >>>
>>> >>> On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> >>>> The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
>>> >>>>
>>> >>>> commit d45db898339e1b2212f5befff1af714abcec034f
>>> >>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> >>>> Date:   Wed Sep 5 13:11:32 2012 -0500
>>> >>>>
>>> >>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>>> >>>> 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>> >>>> 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>> >>>> 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>> >>>> 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>> 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>> >>>> 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>> 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable
>>> >>>> 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>>> >>>> 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> >>>> 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> >>>> 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>>> >>>> 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212 at A-LEG:5060, with session description
>>> >>>> 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>> >>>> 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212 at B-Leg, with session description
>>> >>>> 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>> >>>> 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>> 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>> >>>> 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>> 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description
>>> >>>> 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212 at A-LEG:5060
>>> >>>> 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK
>>> >>>> 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated
>>> >>>> 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 at A-LEG:5060
>>> >>>> 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> >>>> 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> >>>> 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
>>> >>>>
>>> >>>>
>>> >>>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> >>>>
>>> >>>>> update and try again, if it still doesn't work open a jira
>>> >>>>>
>>> >>>>>
>>> >>>>> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> >>>>>> as promised here is the update testing and enabling
>>> >>>>>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to
>>> >>>>>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the
>>> >>>>>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
>>> >>>>>>
>>> >>>>>> I can open a JIRA case on this and provide the console log file / PCAP's ect
>>> >>>>>> if that would help
>>> >>>>>>
>>> >>>>>>
>>> >>>>>> Call Flow with out {sip_wait_for_aleg_ack=true}
>>> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
>>> >>>>>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>> >>>>>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>> >>>>>> sip:13605551212 at B-Leg, with session description
>>> >>>>>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>> >>>>>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>>>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>> >>>>>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>>>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> >>>>>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK
>>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> >>>>>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE
>>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> >>>>>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK
>>> >>>>>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> >>>>>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
>>> >>>>>>
>>> >>>>>> Call Flow with {sip_wait_for_aleg_ack=true} enabled
>>> >>>>>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE
>>> >>>>>> sip:+13605551212 at A-LEG:5060, with session description
>>> >>>>>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying
>>> >>>>>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE
>>> >>>>>> sip:13605551212 at B-Leg, with session description
>>> >>>>>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try
>>> >>>>>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>>>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing
>>> >>>>>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing
>>> >>>>>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212 at B-Leg
>>> >>>>>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session
>>> >>>>>> description
>>> >>>>>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK
>>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> >>>>>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE
>>> >>>>>> sip:+13605551212 at FreeSwitch:5070;transport=udp
>>> >>>>>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK
>>> >>>>>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212 at B-Leg
>>> >>>>>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
>>> >>>>>>
>>> >>>>>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> >>>>>>
>>> >>>>>> No worries I will be out this weekend for the long weekend I will work on
>>> >>>>>> getting the test box upgraded and a test case setup on Tuesday I will report
>>> >>>>>> back the results mid to late next week and provided everything works as I
>>> >>>>>> hope it will I will be happy to pay the Wiki tax :)
>>> >>>>>>
>>> >>>>>>
>>> >>>>>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> >>>>>>
>>> >>>>>>
>>> >>>>>>
>>> >>>>>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame at me.com>
>>> >>>>>> wrote:
>>> >>>>>>>
>>> >>>>>>> Cool I will nail that up on my test box and see if that works
>>> >>>>>>>
>>> >>>>>> Please report back on whether it works or not and then be prepared to pay
>>> >>>>>> the wiki tax. :) I'll be glad to assist with getting this documented
>>> >>>>>> although I think you're in the best position to give that documentation some
>>> >>>>>> real-world context.
>>> >>>>>>
>>> >>>>>> -MC

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From jackal at cybershroud.net  Tue Sep 11 22:43:10 2012
From: jackal at cybershroud.net (Carlos Flor)
Date: Tue, 11 Sep 2012 14:43:10 -0400
Subject: [Freeswitch-users] Polycom Extension Issue
Message-ID: <CAAYhoXJsYL8Wrjc-jz1pPFHGnd75NgYrg88JkNGUxUC8zZp-yA@mail.gmail.com>

So here is the situation:  I have a polycom phone with extension
101 at pbxAand 101 at pbxB. Someone else has a polycom phone with extension
102 at pbxAand 102 at pbxB. If I try to call from 101 at Bto 102 at B,
the phone rings, but as soon as 102 answers, the call hangs up immediately.
 If I change 102 at B to 103 at B, so that the extensions are now 102 at A and
103 at Bon the second phone, then phone calls from the second phone to
the first
work, but not the other way around.  If I repeat the change on the first
phone, so that it now has 101 at A and 104 at B, then calls work correctly in
both directions.

So, it seems as though if you have more than one registration but use the
same extension on each, the polycom has issues with the RTP piece of the
call.  The SIP piece seems to work ok because the call actually makes it to
the other phone and rings, but as soon as you pickup (when RTP should
start) the call ends.

Has anyone run into anything close to this before?  I am sure my
description is confusing and it's much easier to explain on a whiteboard,
but hopefully it makes sense.

Just to clarify, the two PBXs are not related to each other.  I'm not
trying to call from pbxA to pbxB.


Carlos
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From lists at kavun.ch  Tue Sep 11 23:43:34 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 11 Sep 2012 15:43:34 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
Message-ID: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>

Just wanted to drop a quick note about an issue I just stumbled upon.

When using play_and_get_digits, you are still able to input DTMFs when the invalid-entry prompt is played, but the message won't be interrupted and your DTMF input won't be executed until the prompt is fully played.

Can someone look into this and confirm?

Best,
Emrah

From wstephen80 at gmail.com  Wed Sep 12 00:09:22 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 11 Sep 2012 22:09:22 +0200
Subject: [Freeswitch-users] Remove UPDATE from Allow list in the INVITE
Message-ID: <CAO4LeU1UMDY3hNcZPy7VoWSyeRhiTzURTEGy0QU4S+RGofLukg@mail.gmail.com>

It's possible to remove the UPDATE from the Allow list in the INVITE?
I have an issue with a customer where the solution is not to send the
UPDATE in the Allow list.
I have tried with dialplan:

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

without succes.
Any advise?

Stephen
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From anthony.minessale at gmail.com  Wed Sep 12 00:40:24 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 Sep 2012 15:40:24 -0500
Subject: [Freeswitch-users] Remove UPDATE from Allow list in the INVITE
In-Reply-To: <CAO4LeU1UMDY3hNcZPy7VoWSyeRhiTzURTEGy0QU4S+RGofLukg@mail.gmail.com>
References: <CAO4LeU1UMDY3hNcZPy7VoWSyeRhiTzURTEGy0QU4S+RGofLukg@mail.gmail.com>
Message-ID: <CAKbxfG_n4iU4qmvcNf_cWvqkNxE2t7oQVDg3yiWpLvKWfoqv7w@mail.gmail.com>

try the new 1-minute-old send-display-update=false sofia profile param

On Tue, Sep 11, 2012 at 3:09 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> It's possible to remove the UPDATE from the Allow list in the INVITE?
> I have an issue with a customer where the solution is not to send the
> UPDATE in the Allow list.
> I have tried with dialplan:
>
> <action application="export" data="ignore_display_updates=true" />
>
> without succes.
> Any advise?
>
> 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-users mailing list
> FreeSWITCH-users at 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 wstephen80 at gmail.com  Wed Sep 12 01:20:17 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 11 Sep 2012 23:20:17 +0200
Subject: [Freeswitch-users] Remove UPDATE from Allow list in the INVITE
In-Reply-To: <CAKbxfG_n4iU4qmvcNf_cWvqkNxE2t7oQVDg3yiWpLvKWfoqv7w@mail.gmail.com>
References: <CAO4LeU1UMDY3hNcZPy7VoWSyeRhiTzURTEGy0QU4S+RGofLukg@mail.gmail.com>
	<CAKbxfG_n4iU4qmvcNf_cWvqkNxE2t7oQVDg3yiWpLvKWfoqv7w@mail.gmail.com>
Message-ID: <CAO4LeU3iK0R9ruO0tXjRgB3-3pTyzR=9s0f5KuGLPNNoDGvAuA@mail.gmail.com>

Thank you Anthony, I'm updating...

Stephen

On Tue, Sep 11, 2012 at 10:40 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try the new 1-minute-old send-display-update=false sofia profile param
>
> On Tue, Sep 11, 2012 at 3:09 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> It's possible to remove the UPDATE from the Allow list in the INVITE?
>> I have an issue with a customer where the solution is not to send the
>> UPDATE in the Allow list.
>> I have tried with dialplan:
>>
>> <action application="export" data="ignore_display_updates=true" />
>>
>> without succes.
>> Any advise?
>>
>> 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-users mailing list
>> FreeSWITCH-users at 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 msc at freeswitch.org  Wed Sep 12 03:53:42 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Sep 2012 16:53:42 -0700
Subject: [Freeswitch-users] Polycom Extension Issue
In-Reply-To: <CAAYhoXJsYL8Wrjc-jz1pPFHGnd75NgYrg88JkNGUxUC8zZp-yA@mail.gmail.com>
References: <CAAYhoXJsYL8Wrjc-jz1pPFHGnd75NgYrg88JkNGUxUC8zZp-yA@mail.gmail.com>
Message-ID: <CAKzWOxV93ke_UXszssF3BoGZq3MK5CrXX6V-V1=ZtGnSShTNcw@mail.gmail.com>

Can you collect console logs w/ SIP trace? Drop them on
pastebin.freeswitch.org and link back here. I think we have a few Polycom
gurus who may have some experience here, but they probably will need more
info before they can help.

-MC

On Tue, Sep 11, 2012 at 11:43 AM, Carlos Flor <jackal at cybershroud.net>wrote:

>  So here is the situation:  I have a polycom phone with extension 101 at pbxAand 101 at pbxB. Someone else has a polycom phone with extension 102 at pbxAand 102 at pbxB. If I try to call from 101 at Bto 102 at B,
> the phone rings, but as soon as 102 answers, the call hangs up immediately.
>  If I change 102 at B to 103 at B, so that the extensions are now 102 at A and
> 103 at B on the second phone, then phone calls from the second phone to the
> first work, but not the other way around.  If I repeat the change on the
> first phone, so that it now has 101 at A and 104 at B, then calls work
> correctly in both directions.
>
> So, it seems as though if you have more than one registration but use the
> same extension on each, the polycom has issues with the RTP piece of the
> call.  The SIP piece seems to work ok because the call actually makes it to
> the other phone and rings, but as soon as you pickup (when RTP should
> start) the call ends.
>
> Has anyone run into anything close to this before?  I am sure my
> description is confusing and it's much easier to explain on a whiteboard,
> but hopefully it makes sense.
>
> Just to clarify, the two PBXs are not related to each other.  I'm not
> trying to call from pbxA to pbxB.
>
>
> Carlos
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 12 03:56:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Sep 2012 16:56:18 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <9482D8B7-1B61-4EA9-BA45-9473699D1D74@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
Message-ID: <CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>

On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com>wrote:

> Crash was resolved this weekend by a code update via jira case - no more
> crashes after the update - some more dial plan edits and a little more
> testing I will be ready to head over to pay the Wiki tax
>
> Thanks guys for adding this feature it seems to be working as expected and
> my 491 issues on the B-Leg / A-Leg have pretty much disappeared
>

Thanks for checking in and letting us know where you stand. I really like
it when these threads have some sort of resolution. If you have any issues
with wiki editing please let me know or hop on IRC for an assist.

-MC
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From msc at freeswitch.org  Wed Sep 12 03:57:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Sep 2012 16:57:39 -0700
Subject: [Freeswitch-users] Remove UPDATE from Allow list in the INVITE
In-Reply-To: <CAO4LeU3iK0R9ruO0tXjRgB3-3pTyzR=9s0f5KuGLPNNoDGvAuA@mail.gmail.com>
References: <CAO4LeU1UMDY3hNcZPy7VoWSyeRhiTzURTEGy0QU4S+RGofLukg@mail.gmail.com>
	<CAKbxfG_n4iU4qmvcNf_cWvqkNxE2t7oQVDg3yiWpLvKWfoqv7w@mail.gmail.com>
	<CAO4LeU3iK0R9ruO0tXjRgB3-3pTyzR=9s0f5KuGLPNNoDGvAuA@mail.gmail.com>
Message-ID: <CAKzWOxXN-hsFuBmURo67NeOr5csoraU-ZL1kio63x8XS9gGTiQ@mail.gmail.com>

On Tue, Sep 11, 2012 at 2:20 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Thank you Anthony, I'm updating...
>
> Stephen
>

Please let us know what happens! This item needs wikification once we know
it behaves as intended.
-MC
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From msc at freeswitch.org  Wed Sep 12 04:02:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Sep 2012 17:02:23 -0700
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
Message-ID: <CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>

I do not see this behavior on 1.2.stable branch. When I press a key during
the invalid prompt it breaks out of playback immediately and attempts to
process whatever digit(s) I input. I tested with x5000 sample IVR:
Dial 5000
Press 7, wait for "invalid entry" message to start playing
Press 3 while she's saying "that was an invalid entry"
After a few seconds MOH comes on.

-MC

On Tue, Sep 11, 2012 at 12:43 PM, Emrah <lists at kavun.ch> wrote:

> Just wanted to drop a quick note about an issue I just stumbled upon.
>
> When using play_and_get_digits, you are still able to input DTMFs when the
> invalid-entry prompt is played, but the message won't be interrupted and
> your DTMF input won't be executed until the prompt is fully played.
>
> Can someone look into this and confirm?
>
> 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 hexade at hotmail.com  Wed Sep 12 04:29:30 2012
From: hexade at hotmail.com (Adelia C.)
Date: Tue, 11 Sep 2012 20:29:30 -0400
Subject: [Freeswitch-users] FreeSwitch crashes when called party hangs up on
	Prompt Play
Message-ID: <BLU002-W5050F1E45C970CF9EA0A7DA920@phx.gbl>

Problem: 

FreeSwitch crashes when called party hangs up on prompt played (prompt played on called side). 
Interestingly enough, this does not happen in DEV (win32 Windows 2003 VMs) but happens on win32 Windows 2003 HW (real hardware).
Offending lib is libapr.dll. I ran libapr.dl through Dependency Walker on both VM and HW and all is equal. 

FreeSwitch version :  GIT version from 2 weeks ago, Release version only (no Debug on DEV)

Calling Function at crash time (full trace at http://pastebin.freeswitch.org/19876) :

2012-09-11 15:38:09.534355 [NOTICE] switch_cpp.cpp:1227 8b24ba68-79ea-4ab3-9d22-ae3da858f1a1 Proxying Call routes are: {origination_caller_id_number=4152484093,origination_uuid=663ec186-36ed-4f40-8e68-4dad5d8550eb,ignore_early_media=true,group_confirm_key=exec,group_confirm_file=playback //ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav}sofia/gateway/26/4152484093 | [origination_uuid=a9dfe5b9-bdd9-40d9-8a2b-897c5c164825]sofia/gateway/25/4152484093 | [origination_uuid=c7cb97af-1dbb-4c84-81be-c8e794000a27]sofia/gateway/7/4152484093 | [origination_uuid=1612b58e-86e7-45e2-bea3-b81360139fea]sofia/gateway/10/4152484093
6815a8a9-fb0b-41dd-b2d0-cabfe7953ab4 EXECUTE sofia/external/4152484093 at 10.3.220.50 bridge({origination_caller_id_number=4152484093,origination_uuid=663ec186-36ed-4f40-8e68-4dad5d8550eb,ignore_early_media=true,group_confirm_key=exec,group_confirm_file=playback //ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav}sofia/gateway/26/4152484093 | [origination_uuid=a9dfe5b9-bdd9-40d9-8a2b-897c5c164825]sofia/gateway/25/4152484093 | [origination_uuid=c7cb97af-1dbb-4c84-81be-c8e794000a27]sofia/gateway/7/4152484093 | [origination_uuid=1612b58e-86e7-45e2-bea3-b81360139fea]sofia/gateway/10/4152484093)
6815a8a9-fb0b-41dd-b2d0-cabfe7953ab4 2012-09-11 15:38:09.549980 [DEBUG] switch_channel.c:1067 sofia/external/4152484093 at 10.3.220.50 EXPORTING[export_vars] [RFC2822_DATE]=[Tue, 11 Sep 2012 15:38:09 Pacific Daylight Time] to event

Exception:
Event Type:    Error
Event Source:    .NET Runtime
Event Category:    None
Event ID:    1026
Date:        9/11/2012
Time:        3:38:31 PM
User:        N/A
Computer:    ASHSTGSIP02
Description:
Application: FreeSwitchConsole.exe
Framework Version: v4.0.30319
Description: The process was terminated due to an unhandled exception.
Exception Info: System.AccessViolationException
Stack:
   at CoreSession.execute(CoreSession* , SByte* , SByte* )
   at CSharp_CoreSession_Execute(Void* jarg1, SByte* jarg2, SByte* jarg3)
   at FreeSWITCH.Native.freeswitchPINVOKE.CoreSession_Execute(HandleRef jarg1, String jarg2, String jarg3)
   at FreeSWITCH.Native.CoreSession.Execute(String app, String data)
   at CallCloudAppServer.Service.Service.ProxyInboundCall(String rtn, String callerIdNumber, Boolean callRecordingRoute)
   at CallCloudAppServer.Service.CallTracking.CallTrackingService.DoVanillaCallTracking()
   at CallCloudAppServer.Service.CallTracking.CallTrackingService.ProcessIncomingCall()
   at CallCloudAppServer.Core.Dispatcher.Dispatch()
   at CallCloudAppServer.Core.MessageHandler.MessageHandler.ProcessIncomingCall()
   at CallCloudAppServer.Main.Run(AppContext context)
   at FreeSWITCH.AppPluginExecutor.Execute(String args, IntPtr sessionHandle)
   at FreeSWITCH.AppPluginExecutor.Execute(System.String, IntPtr)
   at FreeSWITCH.Loader.Run(System.String, IntPtr)


Cause : passing in wrong parameters to triggers Access Violation - 0xc0000005 - in libapr!apr_socket_sendto.

LOGS and CRASH :
LOGS at http://pastebin.freeswitch.org/19876 
CRASH.DMP at  http://pastebin.freeswitch.org/19877

Any idea? 

Thanks for your help!
A.C.




 		 	   		  
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From casteven at gmail.com  Wed Sep 12 06:38:51 2012
From: casteven at gmail.com (Campbell Steven)
Date: Wed, 12 Sep 2012 14:38:51 +1200
Subject: [Freeswitch-users] UK English Prompt Set Recording
Message-ID: <CAPerZE_XRytyQ25xr0cQ42Dw0M+KVSqF+BT+6G57SGwFEEOCfA@mail.gmail.com>

Hi en_UK FreeSWITCH users,

I'm trying to get a full prompt set (every prompt currently recorded
by Callie including the ztrp prompts etc..) organised. The talent
would be female and prompts would be recorded through GM Voices and
therefore easy to get matching custom recordings for your applications
into the future. The prompt set would then be given to the FreeSWITCH
project for distribution as per the existing prompts.

Anyone who is willing to help me out with funding this or has any
questions please contact me off-list as I'd really like to get this
sorted. Every little bit counts here so even if you just want to drop
a few $ in it all adds up.

Thanks

Campbell


From msc at freeswitch.org  Wed Sep 12 06:47:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Sep 2012 19:47:35 -0700
Subject: [Freeswitch-users] FreeSwitch crashes when called party hangs
 up on Prompt Play
In-Reply-To: <BLU002-W5050F1E45C970CF9EA0A7DA920@phx.gbl>
References: <BLU002-W5050F1E45C970CF9EA0A7DA920@phx.gbl>
Message-ID: <CAKzWOxWJKJmgKuD0J519HkuRBrEhJaSOXh04-=vXPdxbtq6ZQg@mail.gmail.com>

This is excellent information! Could you file it on
jira.freeswitch.orgplease? That will make it easier for the devs to
track it and get it
resolved.

Thanks!
-MC

On Tue, Sep 11, 2012 at 5:29 PM, Adelia C. <hexade at hotmail.com> wrote:

> Problem:
>
> FreeSwitch crashes when called party hangs up on prompt played (prompt
> played on called side).
> Interestingly enough, this does not happen in DEV (win32 Windows 2003 VMs)
> but happens on win32 Windows 2003 HW (real hardware).
> Offending lib is libapr.dll. I ran libapr.dl through Dependency Walker on
> both VM and HW and all is equal.
>
>
> FreeSwitch version :  GIT version from 2 weeks ago, Release version only
> (no Debug on DEV)
>
> Calling Function at crash time (full trace at
> http://pastebin.freeswitch.org/19876) :
>
> 2012-09-11 15:38:09.534355 [NOTICE] switch_cpp.cpp:1227
> 8b24ba68-79ea-4ab3-9d22-ae3da858f1a1 Proxying Call routes are:
> {origination_caller_id_number=4152484093,origination_uuid=663ec186-36ed-4f40-8e68-4dad5d8550eb,ignore_early_media=true,group_confirm_key=exec,group_confirm_file=playback
> //
> ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav}sofia/gateway/26/4152484093<http://ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav%7Dsofia/gateway/26/4152484093>|
> [origination_uuid=a9dfe5b9-bdd9-40d9-8a2b-897c5c164825]sofia/gateway/25/4152484093
> |
> [origination_uuid=c7cb97af-1dbb-4c84-81be-c8e794000a27]sofia/gateway/7/4152484093
> |
> [origination_uuid=1612b58e-86e7-45e2-bea3-b81360139fea]sofia/gateway/10/4152484093
> 6815a8a9-fb0b-41dd-b2d0-cabfe7953ab4 EXECUTE sofia/external/4152484093@
> 10.3.220.50bridge({origination_caller_id_number=4152484093,origination_uuid=663ec186-36ed-4f40-8e68-4dad5d8550eb,ignore_early_media=true,group_confirm_key=exec,group_confirm_file=playback
> //
> ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav}sofia/gateway/26/4152484093<http://ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav%7Dsofia/gateway/26/4152484093>|
> [origination_uuid=a9dfe5b9-bdd9-40d9-8a2b-897c5c164825]sofia/gateway/25/4152484093
> |
> [origination_uuid=c7cb97af-1dbb-4c84-81be-c8e794000a27]sofia/gateway/7/4152484093
> |
> [origination_uuid=1612b58e-86e7-45e2-bea3-b81360139fea]sofia/gateway/10/4152484093)
> 6815a8a9-fb0b-41dd-b2d0-cabfe7953ab4 2012-09-11 15:38:09.549980 [DEBUG]
> switch_channel.c:1067 sofia/external/4152484093 at 10.3.220.50EXPORTING[export_vars] [RFC2822_DATE]=[Tue, 11 Sep 2012 15:38:09 Pacific
> Daylight Time] to event
>
>
> Exception:
>
> Event Type:    Error
> Event Source:    .NET Runtime
> Event Category:    None
> Event ID:    1026
> Date:        9/11/2012
> Time:        3:38:31 PM
> User:        N/A
> Computer:    ASHSTGSIP02
> Description:
> Application: FreeSwitchConsole.exe
> Framework Version: v4.0.30319
> Description: The process was terminated due to an unhandled exception.
> Exception Info: System.AccessViolationException
> Stack:
>    at CoreSession.execute(CoreSession* , SByte* , SByte* )
>    at CSharp_CoreSession_Execute(Void* jarg1, SByte* jarg2, SByte* jarg3)
>    at FreeSWITCH.Native.freeswitchPINVOKE.CoreSession_Execute(HandleRef
> jarg1, String jarg2, String jarg3)
>    at FreeSWITCH.Native.CoreSession.Execute(String app, String data)
>    at CallCloudAppServer.Service.Service.ProxyInboundCall(String rtn,
> String callerIdNumber, Boolean callRecordingRoute)
>    at
> CallCloudAppServer.Service.CallTracking.CallTrackingService.DoVanillaCallTracking()
>    at
> CallCloudAppServer.Service.CallTracking.CallTrackingService.ProcessIncomingCall()
>    at CallCloudAppServer.Core.Dispatcher.Dispatch()
>    at
> CallCloudAppServer.Core.MessageHandler.MessageHandler.ProcessIncomingCall()
>    at CallCloudAppServer.Main.Run(AppContext context)
>    at FreeSWITCH.AppPluginExecutor.Execute(String args, IntPtr
> sessionHandle)
>    at FreeSWITCH.AppPluginExecutor.Execute(System.String, IntPtr)
>    at FreeSWITCH.Loader.Run(System.String, IntPtr)
>
>
>
> Cause : passing in wrong parameters to triggers Access Violation -
> 0xc0000005 - in libapr!apr_socket_sendto.
>
> LOGS and CRASH :
>
> LOGS at http://pastebin.freeswitch.org/19876
> CRASH.DMP at  http://pastebin.freeswitch.org/19877
>
>
> *Any idea? *
>
> Thanks for your help!
> A.C.
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 me at gbraad.nl  Wed Sep 12 08:27:46 2012
From: me at gbraad.nl (@gbraad)
Date: Wed, 12 Sep 2012 12:27:46 +0800
Subject: [Freeswitch-users] Missing RPM repo archive
Message-ID: <CAGrH30UVaUd1JHQfWy1-B_KpHXaQjVXftZnG23hreQaJr6y-_g@mail.gmail.com>

Hi,


The file http://files.freeswitch.org/yum/freeswitch-release-1-0.noarch.rpm
seems to be missing, so for now you would need to create your own repo
file in /etc/yum.repos.d/ as freeswitch.repo containing the following:

[freeswitch]
name=FreeSwitch RPMs for Enterprise Linux $releasever - $basearch
baseurl=http://files.freeswitch.org/yum/$releasever/$basearch/
enabled=1
gpgcheck=0

Take note of the gpgcheck=0, since the key is not available for
verification. Please, provide the needed repo archive...

regards,


Gerard

-- 
Gerard Braad ? ???
   F/OSS & IT Consultant in Beijing
   http://gbraad.nl  gpg: 0x592CFE75


From krice at freeswitch.org  Wed Sep 12 11:13:58 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Sep 2012 02:13:58 -0500
Subject: [Freeswitch-users] Missing RPM repo archive
In-Reply-To: <CAGrH30UVaUd1JHQfWy1-B_KpHXaQjVXftZnG23hreQaJr6y-_g@mail.gmail.com>
Message-ID: <CC75A066.4D007%krice@freeswitch.org>

Wtf,

The file and the verification key should be there....




On 9/11/12 11:27 PM, "@gbraad" <me at gbraad.nl> wrote:

> Hi,


The file 
> http://files.freeswitch.org/yum/freeswitch-release-1-0.noarch.rpm
seems to be
> missing, so for now you would need to create your own repo
file in
> /etc/yum.repos.d/ as freeswitch.repo containing the
> following:

[freeswitch]
name=FreeSwitch RPMs for Enterprise Linux $releasever
> - 
> $basearch
baseurl=http://files.freeswitch.org/yum/$releasever/$basearch/
enabl
> ed=1
gpgcheck=0

Take note of the gpgcheck=0, since the key is not available
> for
verification. Please, provide the needed repo
> archive...

regards,


Gerard

-- 
Gerard Braad ? ???
   F/OSS & IT Consultant
> in Beijing
   http://gbraad.nl  gpg:
> 0x592CFE75

__________________________________________________________________
> _______
Professional 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 rajkumar.kanniappan at sasken.com  Wed Sep 12 11:09:02 2012
From: rajkumar.kanniappan at sasken.com (Rajkumar Kanniappan)
Date: Wed, 12 Sep 2012 12:39:02 +0530
Subject: [Freeswitch-users] IP to PBX calls recovery
Message-ID: <6F91E0FFDA542149961F7BDED2D2B94B69B2E26DFD@EXGMBX01.sasken.com>

Hi,

I'm trying to make connection between IP network and PBX network by using the PRI interface. Also I had succeeded in making the call between IP and PBX.

Now I need to provide the high availability for end users in case of IP failure. I had seen the HA enterprise deployment in case of IP failover which talks about Sofia recovery. Using that I'm able to recover IP to IP calls. My question is how to recover the calls between IP and PBX. Please give your suggestion on this.

Thanks

________________________________
SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email.
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From krice at freeswitch.org  Wed Sep 12 11:17:16 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Sep 2012 02:17:16 -0500
Subject: [Freeswitch-users] Missing RPM repo archive
In-Reply-To: <CAGrH30UVaUd1JHQfWy1-B_KpHXaQjVXftZnG23hreQaJr6y-_g@mail.gmail.com>
Message-ID: <CC75A12C.4D009%krice@freeswitch.org>

Actually that file is at
http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm ...
And the key for the signing is at

http://files.freeswitch.org/yum/RPM-GPG-KEY-FREESWITCH

The Official RPMs are signed ...

Please use the key to verify you have proper RPMs...


On 9/11/12 11:27 PM, "@gbraad" <me at gbraad.nl> wrote:

> Hi,


The file 
> http://files.freeswitch.org/yum/freeswitch-release-1-0.noarch.rpm
seems to be
> missing, so for now you would need to create your own repo
file in
> /etc/yum.repos.d/ as freeswitch.repo containing the
> following:

[freeswitch]
name=FreeSwitch RPMs for Enterprise Linux $releasever
> - 
> $basearch
baseurl=http://files.freeswitch.org/yum/$releasever/$basearch/
enabl
> ed=1
gpgcheck=0

Take note of the gpgcheck=0, since the key is not available
> for
verification. Please, provide the needed repo
> archive...

regards,


Gerard

-- 
Gerard Braad ? ???
   F/OSS & IT Consultant
> in Beijing
   http://gbraad.nl  gpg:
> 0x592CFE75

__________________________________________________________________
> _______
Professional 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 hexade at hotmail.com  Wed Sep 12 11:22:54 2012
From: hexade at hotmail.com (Adelia C.)
Date: Wed, 12 Sep 2012 03:22:54 -0400
Subject: [Freeswitch-users] FreeSwitch crashes when called party hangs
 up on Prompt Play
In-Reply-To: <CAKzWOxWJKJmgKuD0J519HkuRBrEhJaSOXh04-=vXPdxbtq6ZQg@mail.gmail.com>
References: <BLU002-W5050F1E45C970CF9EA0A7DA920@phx.gbl>,
	<CAKzWOxWJKJmgKuD0J519HkuRBrEhJaSOXh04-=vXPdxbtq6ZQg@mail.gmail.com>
Message-ID: <BLU002-W140E9D363A2ECB251AE9238DA920@phx.gbl>

Thank you Michael. Ticket is 
FS-4612 - FreeSwitch crashes when called party hangs up on Prompt Play? .
Adelia

Date: Tue, 11 Sep 2012 19:47:35 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch crashes when called party hangs up on Prompt Play

This is excellent information! Could you file it on jira.freeswitch.org please? That will make it easier for the devs to track it and get it resolved.

Thanks!
-MC


On Tue, Sep 11, 2012 at 5:29 PM, Adelia C. <hexade at hotmail.com> wrote:




Problem: 

FreeSwitch crashes when called party hangs up on prompt played (prompt played on called side). 
Interestingly enough, this does not happen in DEV (win32 Windows 2003 VMs) but happens on win32 Windows 2003 HW (real hardware).

Offending lib is libapr.dll. I ran libapr.dl through Dependency Walker on both VM and HW and all is equal. 

FreeSwitch version :  GIT version from 2 weeks ago, Release version only (no Debug on DEV)


Calling Function at crash time (full trace at http://pastebin.freeswitch.org/19876) :

2012-09-11 15:38:09.534355 [NOTICE] switch_cpp.cpp:1227 8b24ba68-79ea-4ab3-9d22-ae3da858f1a1 Proxying Call routes are: {origination_caller_id_number=4152484093,origination_uuid=663ec186-36ed-4f40-8e68-4dad5d8550eb,ignore_early_media=true,group_confirm_key=exec,group_confirm_file=playback //ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav}sofia/gateway/26/4152484093 | [origination_uuid=a9dfe5b9-bdd9-40d9-8a2b-897c5c164825]sofia/gateway/25/4152484093 | [origination_uuid=c7cb97af-1dbb-4c84-81be-c8e794000a27]sofia/gateway/7/4152484093 | [origination_uuid=1612b58e-86e7-45e2-bea3-b81360139fea]sofia/gateway/10/4152484093

6815a8a9-fb0b-41dd-b2d0-cabfe7953ab4 EXECUTE sofia/external/4152484093 at 10.3.220.50 bridge({origination_caller_id_number=4152484093,origination_uuid=663ec186-36ed-4f40-8e68-4dad5d8550eb,ignore_early_media=true,group_confirm_key=exec,group_confirm_file=playback //ashstgfs.stage.keen.com/files/Callcloud/NewWhisper_d5d694a0_a402_491e_a5c2_ac15c8c5b92f.wav}sofia/gateway/26/4152484093 | [origination_uuid=a9dfe5b9-bdd9-40d9-8a2b-897c5c164825]sofia/gateway/25/4152484093 | [origination_uuid=c7cb97af-1dbb-4c84-81be-c8e794000a27]sofia/gateway/7/4152484093 | [origination_uuid=1612b58e-86e7-45e2-bea3-b81360139fea]sofia/gateway/10/4152484093)

6815a8a9-fb0b-41dd-b2d0-cabfe7953ab4 2012-09-11 15:38:09.549980 [DEBUG] switch_channel.c:1067 sofia/external/4152484093 at 10.3.220.50 EXPORTING[export_vars] [RFC2822_DATE]=[Tue, 11 Sep 2012 15:38:09 Pacific Daylight Time] to event


Exception:
Event Type:    Error
Event Source:    .NET Runtime
Event Category:    None
Event ID:    1026
Date:        9/11/2012
Time:        3:38:31 PM

User:        N/A
Computer:    ASHSTGSIP02
Description:
Application: FreeSwitchConsole.exe
Framework Version: v4.0.30319
Description: The process was terminated due to an unhandled exception.
Exception Info: System.AccessViolationException

Stack:
   at CoreSession.execute(CoreSession* , SByte* , SByte* )
   at CSharp_CoreSession_Execute(Void* jarg1, SByte* jarg2, SByte* jarg3)
   at FreeSWITCH.Native.freeswitchPINVOKE.CoreSession_Execute(HandleRef jarg1, String jarg2, String jarg3)

   at FreeSWITCH.Native.CoreSession.Execute(String app, String data)
   at CallCloudAppServer.Service.Service.ProxyInboundCall(String rtn, String callerIdNumber, Boolean callRecordingRoute)
   at CallCloudAppServer.Service.CallTracking.CallTrackingService.DoVanillaCallTracking()

   at CallCloudAppServer.Service.CallTracking.CallTrackingService.ProcessIncomingCall()
   at CallCloudAppServer.Core.Dispatcher.Dispatch()
   at CallCloudAppServer.Core.MessageHandler.MessageHandler.ProcessIncomingCall()

   at CallCloudAppServer.Main.Run(AppContext context)
   at FreeSWITCH.AppPluginExecutor.Execute(String args, IntPtr sessionHandle)
   at FreeSWITCH.AppPluginExecutor.Execute(System.String, IntPtr)
   at FreeSWITCH.Loader.Run(System.String, IntPtr)



Cause : passing in wrong parameters to triggers Access Violation - 0xc0000005 - in libapr!apr_socket_sendto.

LOGS and CRASH :
LOGS at http://pastebin.freeswitch.org/19876 

CRASH.DMP at  http://pastebin.freeswitch.org/19877

Any idea? 

Thanks for your help!

A.C.




 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at 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
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From gvvsubhashkumar at gmail.com  Wed Sep 12 13:21:40 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Wed, 12 Sep 2012 02:21:40 -0700
Subject: [Freeswitch-users] Bridge after Leg B has connected
Message-ID: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>

Hi,

 I have a simple dialplan that just do bridge for incoming calls. I'm
trying to do bridge the Call Leg A with Call Leg B after getting
confirmation of call connected from Call Leg B

  <action application="bridge" data="(sofia or
freetdm)/internal/destination number"/> **

Is there any way to bridge the both the call legs after getting
confirmation from call leg B?


Thanks,
Subhash.
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From avi at avimarcus.net  Wed Sep 12 13:40:04 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 12 Sep 2012 12:40:04 +0300
Subject: [Freeswitch-users] Bridge after Leg B has connected
In-Reply-To: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
References: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
Message-ID: <CALNZuwmMQ8nhgjk=+RFQ516PRZCjXfoNSUzmAKvMOo4K2Mf2qQ@mail.gmail.com>

You want to create a 3 way call once the 2nd party picks up?
-Avi


On Wed, Sep 12, 2012 at 12:21 PM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi,
>
>  I have a simple dialplan that just do bridge for incoming calls. I'm
> trying to do bridge the Call Leg A with Call Leg B after getting
> confirmation of call connected from Call Leg B
>
>   <action application="bridge" data="(sofia or
> freetdm)/internal/destination number"/> **
>
> Is there any way to bridge the both the call legs after getting
> confirmation from call leg B?
>
>
> 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
>
>
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From gvvsubhashkumar at gmail.com  Wed Sep 12 13:50:28 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Wed, 12 Sep 2012 02:50:28 -0700
Subject: [Freeswitch-users] Bridge after Leg B has connected
In-Reply-To: <CALNZuwmMQ8nhgjk=+RFQ516PRZCjXfoNSUzmAKvMOo4K2Mf2qQ@mail.gmail.com>
References: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
	<CALNZuwmMQ8nhgjk=+RFQ516PRZCjXfoNSUzmAKvMOo4K2Mf2qQ@mail.gmail.com>
Message-ID: <CAFvodxVJ-1qS3Rnuo+ZBWSFe8UpwX31cfcLH-ft0_+hVydAfQw@mail.gmail.com>

No i want to capture the reason code if leg B is not connected


Thanks,
Subhash.


On Wed, Sep 12, 2012 at 2:40 AM, Avi Marcus <avi at avimarcus.net> wrote:

> You want to create a 3 way call once the 2nd party picks up?
> -Avi
>
>
> On Wed, Sep 12, 2012 at 12:21 PM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>
>> Hi,
>>
>>  I have a simple dialplan that just do bridge for incoming calls. I'm
>> trying to do bridge the Call Leg A with Call Leg B after getting
>> confirmation of call connected from Call Leg B
>>
>>   <action application="bridge" data="(sofia or
>> freetdm)/internal/destination number"/> **
>>
>> Is there any way to bridge the both the call legs after getting
>> confirmation from call leg B?
>>
>>
>> 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 vallimamod.abdullah at imtelecom.fr  Wed Sep 12 16:04:11 2012
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Wed, 12 Sep 2012 14:04:11 +0200
Subject: [Freeswitch-users] sip-router or opensips for good
In-Reply-To: <503DF591.6040803@pocock.com.au>
References: <mailman.17877.1346234570.13443.freeswitch-users@lists.freeswitch.org>
	<503DEC61.10607@redembedded.com> <503DF591.6040803@pocock.com.au>
Message-ID: <4C5F4D9C-747D-4F01-B5BB-95B680A6D9EC@imtelecom.fr>

Hi,

This is very interesting and to dig further, what is the best use of the proxy: should it be only used as a loadbalancer and let freeswitch handle the registrations or should it handle the registrations too? I have only seen the first configuration on the wiki or ML so far.

In both cases, how do you share the registration information between the front proxy and the ipbx ? And how do you manage failover ?

Best Regards,
- vallimamod.



On Aug 29, 2012, at 12:57 PM, Daniel Pocock <daniel at pocock.com.au> wrote:

> 
> 
> 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.



From tech at fahrwerk-berlin.de  Wed Sep 12 16:37:56 2012
From: tech at fahrwerk-berlin.de (Fahrwerk Kurierkollektiv (Technik))
Date: Wed, 12 Sep 2012 14:37:56 +0200
Subject: [Freeswitch-users] Callers are put into queue although there are
	free phones
Message-ID: <50508224.1060401@fahrwerk-berlin.de>

Hey,

we're running a freeswitch instance in our small company on a Linux HA 
Cluster under Debian. Since we are a bicycle messenger company, a 
correctly working telephone system is absolutely crucial.
We use a total of four SNOM Phones (300 series), two of which are 
"incoming" for calling customers and the remaining two are used to 
communicate with our couriers, e.g. in case that the radio doesn't work.

Of lately, some customers reported to us, that they have been put into 
our queue for more than 10 minutes until they hang up. A normal call 
from or to a customer is not longer than 4 minutes and in most cases we 
are sure that there was absolutely no reason to put the person into the 
queue because both phones were free at the time of calling.

A collegue of mine proposed that it might have to do something with the 
following advice taken from the wiki:

  "If calls are not properly ended by freeswitch after the phone has
   hung up, check if the BYE messages sent by the phone are
   authenticated. (containing a proxy-authentication line) If not, you
   can work around this by adding <param name="auth-all-packets"
   value="false"/> to the sip-profile."

http://wiki.freeswitch.org/wiki/Interop_List#SNOM_190.2C_snom300_series.2C_snom800_series.2C_snomMP.2C_snomM9.2C_snomPA1
(The callers number is 493027574754)

I have posted a freeswitch log to pastebin of one confirmed call where 
the customer was put into queue although none of the phones was occupied:
http://pastebin.com/NdVZ9DdS

Also here's how we set up our queue, which is fairly simple I guess:
http://pastebin.com/sGNBmk2d

I can, of course, provide more configuration snippets and so on, if 
requested and hope that you can help us fix the problem :)

Greets,
Markus


From tech at fahrwerk-berlin.de  Wed Sep 12 16:45:46 2012
From: tech at fahrwerk-berlin.de (Fahrwerk Kurierkollektiv (Technik))
Date: Wed, 12 Sep 2012 14:45:46 +0200
Subject: [Freeswitch-users] Callers are put into queue although there
 are free phones
In-Reply-To: <50508224.1060401@fahrwerk-berlin.de>
References: <50508224.1060401@fahrwerk-berlin.de>
Message-ID: <505083FA.7000300@fahrwerk-berlin.de>

Hey,

> I have posted a freeswitch log to pastebin of one confirmed call where
> the customer was put into queue although none of the phones was occupied:
> http://pastebin.com/NdVZ9DdS
 > (The callers number is 493027574754)

Fixed. Sorry for double-posting. I thought it might need the clarification.

Greets,
Markus



From xyangni at gmail.com  Wed Sep 12 17:18:58 2012
From: xyangni at gmail.com (Yihui Li)
Date: Wed, 12 Sep 2012 14:18:58 +0100
Subject: [Freeswitch-users] GSMOPEN to send long SMS
Message-ID: <CAO1DH1_1CW8SWpV1BrZUHByg9GsbbMfgeY7jp=ig8RbCUQws1A@mail.gmail.com>

Hi,

I have used mod_gsmopen to send short text messages from my E1550. However,
where the length is longer than 160 chars, it always fail. Is there any
solution to send long sms like the mobile phone?

Thanks,
Eric
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From ben at langfeld.co.uk  Wed Sep 12 17:37:51 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 12 Sep 2012 14:37:51 +0100
Subject: [Freeswitch-users] GSMOPEN to send long SMS
In-Reply-To: <CAO1DH1_1CW8SWpV1BrZUHByg9GsbbMfgeY7jp=ig8RbCUQws1A@mail.gmail.com>
References: <CAO1DH1_1CW8SWpV1BrZUHByg9GsbbMfgeY7jp=ig8RbCUQws1A@mail.gmail.com>
Message-ID: <CAAyX+KHgZGmLWykAdfXg=ry62ossNRtZb3Rts70u=Byq3QY4ZQ@mail.gmail.com>

Send the message in multiple parts of <= 160 chars each?

Regards,
Ben Langfeld


On 12 September 2012 14:18, Yihui Li <xyangni at gmail.com> wrote:

> Hi,
>
> I have used mod_gsmopen to send short text messages from my E1550.
> However, where the length is longer than 160 chars, it always fail. Is
> there any solution to send long sms like the mobile phone?
>
> Thanks,
> Eric
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From carl-johan at seacom.dk  Wed Sep 12 16:21:27 2012
From: carl-johan at seacom.dk (Carl Johan Jensen)
Date: Wed, 12 Sep 2012 14:21:27 +0200
Subject: [Freeswitch-users] link error when compiling mod_dingaling
Message-ID: <010101cd90e1$21db8b90$6592a2b0$@dk>

I have cloed the V1.2.stable and tries to compile it using Visual C++ 2008

I get the below linking error (and several more)

What am I missing to include / do ?

 

3>   Creating library Win32\Debug/mod_dingaling.2008.lib and object
Win32\Debug/mod_dingaling.2008.exp

3>iksemel.lib(stream.obj) : error LNK2019: unresolved external symbol
_SSL_free referenced in function _handshake

3>iksemel.lib(stream.obj) : error LNK2019: unresolved external symbol
_SSL_connect referenced in function _handshake

 

Best Regards

Carl Johan Jensen

 

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From manish.kumar.fs at gmail.com  Wed Sep 12 17:33:57 2012
From: manish.kumar.fs at gmail.com (Manish Kumar)
Date: Wed, 12 Sep 2012 19:03:57 +0530
Subject: [Freeswitch-users] Bridge after Leg B has connected
In-Reply-To: <CAFvodxVJ-1qS3Rnuo+ZBWSFe8UpwX31cfcLH-ft0_+hVydAfQw@mail.gmail.com>
References: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
	<CALNZuwmMQ8nhgjk=+RFQ516PRZCjXfoNSUzmAKvMOo4K2Mf2qQ@mail.gmail.com>
	<CAFvodxVJ-1qS3Rnuo+ZBWSFe8UpwX31cfcLH-ft0_+hVydAfQw@mail.gmail.com>
Message-ID: <50508F45.2060908@gmail.com>

On 9/12/2012 3:20 PM, Subhash wrote:
> No i want to capture the reason code if leg B is not connected
>
>
> Thanks,
> Subhash.
>
>
> On Wed, Sep 12, 2012 at 2:40 AM, Avi Marcus <avi at avimarcus.net 
> <mailto:avi at avimarcus.net>> wrote:
>
>     You want to create a 3 way call once the 2nd party picks up?
>     -Avi
>
>
>     On Wed, Sep 12, 2012 at 12:21 PM, Subhash
>     <gvvsubhashkumar at gmail.com <mailto:gvvsubhashkumar at gmail.com>> wrote:
>
>         Hi,
>
>          I have a simple dialplan that just do bridge for incoming
>         calls. I'm trying to do bridge the Call Leg A with Call Leg B
>         after getting confirmation of call connected from Call Leg B
>
>           <action application="bridge" data="(sofia or
>         freetdm)/internal/destination number"/>
>
>         Is there any way to bridge the both the call legs after
>         getting confirmation from call leg B?
>
>
>         Thanks,
>         Subhash.
>
>
>         _________________________________________________________________________
>         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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Try This, it returns the hangup cause of the bridge application via 
channel variable "bridge_hangup_cause"
<action application="log" data="1 B-leg hangup cause: 
${bridge_hangup_cause}"/>
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From daveh at beachdognet.com  Wed Sep 12 17:41:16 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Wed, 12 Sep 2012 09:41:16 -0400
Subject: [Freeswitch-users] channels not in sync on stereo recording
Message-ID: <k2q3dr$q09$1@ger.gmane.org>

I'm experiencing a strange problem on a recording of an audio stream 
where the two channels (I am recording in stereo) are not in sync.  By 
that, I mean when I play back the audio one of the tracks is somehow 
time-shifted, so that it doesn't match up to the proper place in the 
timeline with the other side.  For instance, if I call out to a bank 
IVR system that plays a prompt, and I barge into the prompt by pressing 
a DTMF key, on the recording I will hear the prompt start and then stop 
at the same place it stopped when I pressed the DTMF, but the DTMF key 
itself does not show up on the recording for 5 seconds more.  Has 
anyone ever heard of anything like this?

Now, a few more details.  I am creating a bridge conference and then I 
am recording the audio stream to and from the called party and the 
conference.  I do not have this problem if I record the audio stream to 
and from the calling party and the conference.

Here is the dial plan I am using:

    <extension name="6174973895" continue="false">
      <condition field="source" expression="mod_sofia"/>
      <condition field="destination_number" expression="6174973895">
                <action application="ring_ready"/>
                <action application="answer"/>
                <action application="start_dtmf"/>
                <action application="set" 
data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
                <action application="export" 
data="outfile=$${base_dir}/tmp/${call_id}.wav"/>

                <action application="lua" data="demo_call2.lua"/>

                <action application="export" data="RECORD_STEREO=true"/>
                <action application="export" 
data="nolocal:execute_on_media=record_session ${outfile}"/>

                <action application="export" 
data="nolocal:api_on_answer=conference ${call_id} play 
misc/this-call-may-be-recorded.wav async"/>
                <action application="conference" data="bridge: 
${call_id}@simple:{origination_caller_id_number=${caller_id_number}}sofia/normal_customer/${outdial}@${egress_gateway}"/> 

        </condition>
    </extension>

Any thoughts on what might be going on?




From iam at onnet.su  Wed Sep 12 17:53:14 2012
From: iam at onnet.su (iam)
Date: Wed, 12 Sep 2012 06:53:14 -0700 (PDT)
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <CANBhyPB2dZuAqG3CraDP=Hv8VU-c_MbF8AMOuruVpicQh_5YyQ@mail.gmail.com>
References: <4F87EF60.3070105@vts24.ru>
	<CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
	<4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru>
	<CAKzWOxVs4qWH2ECOORXDQob5AOCrznitJabvLj+N+GGZ4Dzjig@mail.gmail.com>
	<4FAB719E.9010800@vts24.ru> <4FB4C69A.7070108@vts24.ru>
	<CANBhyPB2dZuAqG3CraDP=Hv8VU-c_MbF8AMOuruVpicQh_5YyQ@mail.gmail.com>
Message-ID: <1347457994946-7582782.post@n2.nabble.com>

Hi Vitaly!

Did you ever solve this problem?
I have the same continue_on_fail behaviour.

http://jira.freeswitch.org/browse/FS-4232

Vitaly added a comment - 26/Jun/12 3:14 AM
Sorry for the long absence, but the problem is still actual for me.
I replaced in my dialplan continue_on_fail to "true", but it's not give
desired effect.
May be some additional variables can help me?

Regards,
Kirill



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From zen at freedbms.net  Wed Sep 12 17:59:55 2012
From: zen at freedbms.net (Zenaan Harkness)
Date: Wed, 12 Sep 2012 23:59:55 +1000
Subject: [Freeswitch-users] Note: ./configure with --enable-64 on 32-bit
 platform gives "error: no usable zlib;
 please install zlib devel package or equivalent"
Message-ID: <CAOsGNST8bb5+caR-PVZfwkue4+a2_cTahO3z2XyG8Dd1Venr4g@mail.gmail.com>

As per subject, ./configure when run with --enable-64 on 32-bit Debian
testing gives me "error: no usable zlib; please install zlib devel
package or equivalent".

Perhaps it would be good to have a ./configure warning or something -
I guess it's conceivable that someone might want to cross compile _to_
amd64, _from_ a 32-bit platform (is this even possible?) but I'd
assume would generally be an error.

This email was going to be a plea for help/ assumed bug, before I
(thankfully, and saving me from a brown paper bag) realised I was
copying my 64-bit command line into my 32-bit computer's remote
session ...

The erroneous (from my perspective) output had me try a few pathways
over the last day+ (yeah yeah, brown paper bag), before I just then
realised (don't rub it in) that --enable-64 is perhaps not quite ideal
on 32 bit hardware.

(Aside: for the record, I've tried testing freeswitch and client on
same (dual core) laptop:
linphone won't log in (sip) and only says "port 5060 may already be in
use" or somesuch" - perhaps it's trying to run as a server?;
jitsi works but really rough latency, huge mouse lags and jumps etc
(yet to test with ..-rt real time debian kernel);
qutecom appears to work very well so far (5s audio echo and space
invaders soundtrack tests).
)

zen


From x.liu at hw.ac.uk  Wed Sep 12 18:20:45 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Wed, 12 Sep 2012 15:20:45 +0100
Subject: [Freeswitch-users] Is the FS JIRA server not working properly now?
In-Reply-To: <CAMwZ-tLvETVM5VtgovcX-CXBfVSdzoN3bVu3QYEwvqsTzxaM8g@mail.gmail.com>
References: <20120910031044.GE15903@ztillma.com>
	<CAMwZ-tLvETVM5VtgovcX-CXBfVSdzoN3bVu3QYEwvqsTzxaM8g@mail.gmail.com>
Message-ID: <50509A3D.7010500@hw.ac.uk>

Hi,

Normally after I add a comment to an existing issue I will receive an 
email copy of the comment very soon.

I haven't received the email after I added a comment 6 hours ago today.
so I am wondering if there is something wrong.

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.



From jack at livecall.com  Wed Sep 12 18:49:48 2012
From: jack at livecall.com (Jack)
Date: Wed, 12 Sep 2012 07:49:48 -0700
Subject: [Freeswitch-users] Work around for Chrome and Flex Phone
Message-ID: <5050A10C.1020904@livecall.com>

Chrome can be tweaked to work with the Flex phone on 
conference.freeswitch.org by going to the plugins ( chrome://plugins ) 
and disabling pepflashplayer.dll.
Chrome Canary works without tweaking.


From msc at freeswitch.org  Wed Sep 12 18:58:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Sep 2012 07:58:39 -0700
Subject: [Freeswitch-users] FreeSwitch crashes when called party hangs
 up on Prompt Play
In-Reply-To: <BLU002-W140E9D363A2ECB251AE9238DA920@phx.gbl>
References: <BLU002-W5050F1E45C970CF9EA0A7DA920@phx.gbl>
	<CAKzWOxWJKJmgKuD0J519HkuRBrEhJaSOXh04-=vXPdxbtq6ZQg@mail.gmail.com>
	<BLU002-W140E9D363A2ECB251AE9238DA920@phx.gbl>
Message-ID: <CAKzWOxXTxdFaSxCqmbRELQAaXJia-6EJ56gY3HjZsSGuRJC9Qg@mail.gmail.com>

On Wed, Sep 12, 2012 at 12:22 AM, Adelia C. <hexade at hotmail.com> wrote:

> Thank you Michael. Ticket is  FS-4612 - FreeSwitch crashes when called
> party hangs up on Prompt Play? <http://jira.freeswitch.org/browse/FS-4612>
>  .
>
> Adelia
>
Thanks for doing that. :)
-MC
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From msc at freeswitch.org  Wed Sep 12 19:02:07 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Sep 2012 08:02:07 -0700
Subject: [Freeswitch-users] sip-router or opensips for good
In-Reply-To: <4C5F4D9C-747D-4F01-B5BB-95B680A6D9EC@imtelecom.fr>
References: <mailman.17877.1346234570.13443.freeswitch-users@lists.freeswitch.org>
	<503DEC61.10607@redembedded.com> <503DF591.6040803@pocock.com.au>
	<4C5F4D9C-747D-4F01-B5BB-95B680A6D9EC@imtelecom.fr>
Message-ID: <CAKzWOxW8R3bDm_4HeKPEEgb2tQ8K8OpHzH86KnWpgp1u3uvwFA@mail.gmail.com>

For the record, Daniel and Scott will be joining us next week and the
following Wednesday to discuss repro. Be sure to call in every Wednesday at
GMT 1700 (1PM EDT, 10AM PDT) to see what interesting things we are
discussing.

-MC

On Wed, Sep 12, 2012 at 5:04 AM, Vallimamod ABDULLAH <
vallimamod.abdullah at imtelecom.fr> wrote:

> Hi,
>
> This is very interesting and to dig further, what is the best use of the
> proxy: should it be only used as a loadbalancer and let freeswitch handle
> the registrations or should it handle the registrations too? I have only
> seen the first configuration on the wiki or ML so far.
>
> In both cases, how do you share the registration information between the
> front proxy and the ipbx ? And how do you manage failover ?
>
> Best Regards,
> - vallimamod.
>
>
>
> On Aug 29, 2012, at 12:57 PM, Daniel Pocock <daniel at pocock.com.au> wrote:
>
> >
> >
> > 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.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 12 19:09:06 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Sep 2012 08:09:06 -0700
Subject: [Freeswitch-users] Bridge after Leg B has connected
In-Reply-To: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
References: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
Message-ID: <CAKzWOxUgXFMTEVH_6Zx55Ga2CSQNmQuEsLPOFpGBan85MM3Z4g@mail.gmail.com>

On Wed, Sep 12, 2012 at 2:21 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi,
>
>  I have a simple dialplan that just do bridge for incoming calls. I'm
> trying to do bridge the Call Leg A with Call Leg B after getting
> confirmation of call connected from Call Leg B
>
>   <action application="bridge" data="(sofia or
> freetdm)/internal/destination number"/> **
>
> Is there any way to bridge the both the call legs after getting
> confirmation from call leg B?
>
The legs will be bridged as soon as the B leg returns media (SIP 180/183)
or in the case of ignoring early media it bridges when receiving a SIP 200
OK. In other words, the bridge app does all the work for you.

Now, if you want the called party to press a key to acknowledge receipt of
the call then you need the 'answer confirmation' feature:
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

HtH,
MC

>
> 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
>
>


-- 
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 Sep 12 19:11:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Sep 2012 08:11:46 -0700
Subject: [Freeswitch-users] IP to PBX calls recovery
In-Reply-To: <6F91E0FFDA542149961F7BDED2D2B94B69B2E26DFD@EXGMBX01.sasken.com>
References: <6F91E0FFDA542149961F7BDED2D2B94B69B2E26DFD@EXGMBX01.sasken.com>
Message-ID: <CAKzWOxXLapfP+EXdfXyLO1C8E-q9-3MaizoNieSVTeqHw31bSQ@mail.gmail.com>

Most likely this will require more than a hint from a friendly community
member. I suggest contacting consulting at freeswitch.org and seeking
professional assistance.
-MC

On Wed, Sep 12, 2012 at 12:09 AM, Rajkumar Kanniappan <
rajkumar.kanniappan at sasken.com> wrote:

>  Hi, ****
>
> ** **
>
> I'm trying to make connection between IP network and PBX network by using
> the PRI interface. Also I had succeeded in making the call between IP and
> PBX.****
>
> ** **
>
> Now I need to provide the high availability for end users in case of IP
> failure. I had seen the HA enterprise deployment in case of IP failover
> which talks about Sofia recovery. Using that I'm able to recover IP to IP
> calls. My question is how to recover the calls between IP and PBX. Please
> give your suggestion on this.****
>
> ** **
>
> Thanks****
>
> ------------------------------
> SASKEN BUSINESS DISCLAIMER: This message may contain confidential,
> proprietary or legally privileged information. In case you are not the
> original intended Recipient of the message, you must not, directly or
> indirectly, use, disclose, distribute, print, or copy any part of this
> message and you are requested to delete it and inform the sender. Any views
> expressed in this message are those of the individual sender unless
> otherwise stated. Nothing contained in this message shall be construed as
> an offer or acceptance of any offer by Sasken Communication Technologies
> Limited ("Sasken") unless sent with that express intent and with due
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> caused by any virus transmitted by this email.
> Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jackal at cybershroud.net  Wed Sep 12 19:12:28 2012
From: jackal at cybershroud.net (Carlos Flor)
Date: Wed, 12 Sep 2012 11:12:28 -0400
Subject: [Freeswitch-users] Polycom Extension Issue
In-Reply-To: <CAKzWOxV93ke_UXszssF3BoGZq3MK5CrXX6V-V1=ZtGnSShTNcw@mail.gmail.com>
References: <CAAYhoXJsYL8Wrjc-jz1pPFHGnd75NgYrg88JkNGUxUC8zZp-yA@mail.gmail.com>
	<CAKzWOxV93ke_UXszssF3BoGZq3MK5CrXX6V-V1=ZtGnSShTNcw@mail.gmail.com>
Message-ID: <CAAYhoXKMyccj9whwS5gdh+t5kJhFgh=E=Of5Onmn2X6pec9wow@mail.gmail.com>

Ok, I have posted a siptrace of the call failing.  If you need more
information, just let me know and I'll get what I can.  I cleaned up domain
names and CallerID names, but everything else should be untouched.  The
link is
http://pastebin.freeswitch.org/19886

Thanks again.

Carlos

On Tue, Sep 11, 2012 at 7:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Can you collect console logs w/ SIP trace? Drop them on
> pastebin.freeswitch.org and link back here. I think we have a few Polycom
> gurus who may have some experience here, but they probably will need more
> info before they can help.
>
> -MC
>
> On Tue, Sep 11, 2012 at 11:43 AM, Carlos Flor <jackal at cybershroud.net>wrote:
>
>>  So here is the situation:  I have a polycom phone with extension
>> 101 at pbxA and 101 at pbxB.  Someone else has a polycom phone with extension
>> 102 at pbxA and 102 at pbxB.  If I try to call from 101 at B to 102 at B, the phone
>> rings, but as soon as 102 answers, the call hangs up immediately.  If I
>> change 102 at B to 103 at B, so that the extensions are now 102 at A and 103 at B on
>> the second phone, then phone calls from the second phone to the first work,
>> but not the other way around.  If I repeat the change on the first phone,
>> so that it now has 101 at A and 104 at B, then calls work correctly in both
>> directions.
>>
>> So, it seems as though if you have more than one registration but use the
>> same extension on each, the polycom has issues with the RTP piece of the
>> call.  The SIP piece seems to work ok because the call actually makes it to
>> the other phone and rings, but as soon as you pickup (when RTP should
>> start) the call ends.
>>
>> Has anyone run into anything close to this before?  I am sure my
>> description is confusing and it's much easier to explain on a whiteboard,
>> but hopefully it makes sense.
>>
>> Just to clarify, the two PBXs are not related to each other.  I'm not
>> trying to call from pbxA to pbxB.
>>
>>
>> Carlos
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Wed Sep 12 19:22:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Sep 2012 08:22:39 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxUzWdm+_04x=3zgKkPiXb75biQrGmoYYm5cf87GpSx1Yw@mail.gmail.com>

Hello All,

Our FreeSWITCH conference call will be taking place in a bit. The agenda is
here:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_12

We look forward to hearing from Ben and Ben about the Adhearsion project.
We also have a few community updates.

-- 
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  Wed Sep 12 19:23:45 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Wed, 12 Sep 2012 08:23:45 -0700 (PDT)
Subject: [Freeswitch-users] More information on FreeSwitch
In-Reply-To: <CAKzWOxX3gV9GABaqWbZ6_=jRPOWXxk-fDM+KMnGRe3wx7b890w@mail.gmail.com>
References: <CC20783F-329B-4377-914E-CE4D640529C6@gmail.com>
	<CACc3D0XO5kOZ_imMy6XazzTGyYUpgJtrRRoAtkK9_WKTbFFo+w@mail.gmail.com>
	<CALNZuwnv1SzQ8FNC47U7SZuwAFo7u6JK3mxaGTmmOkVOfMF=rQ@mail.gmail.com>
	<1347034731293-7582678.post@n2.nabble.com>
	<CAKzWOxX3gV9GABaqWbZ6_=jRPOWXxk-fDM+KMnGRe3wx7b890w@mail.gmail.com>
Message-ID: <1347463425114-7582790.post@n2.nabble.com>

>This is one reason we are doing a second edition to the "bridge book." We
are planning on having an entire chapter dedicated just to handling NAT.

That is excellent news.  I'll definitely be looking forward to that, thanks
for the heads up.  In the meantime, are there any plans to add far-end
NAT-traversal functionality to FS or is it already there and I just need to
figure out how to make it work?  In other words:
FS | <Internet> -- NAT firewall | SIP phone A, SIP phone B, etc. on the same
LAN

Phone A calls Phone B and FS tries to route the media from Phone A to Phone
B using their external NAT addresses (which doesn't work) instead of just
sending the media between the two phones using their internal addresses. 
Proxying the media through FS works but wastes bandwidth and adds latency
when the media could just be going between the two phones.

Can FS handle this or is an SBC necessary?  


>Thanks for the info and feedback!

No problem... I just thought you guys should know about it.

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-information-on-FreeSwitch-tp7582648p7582790.html
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From gvvsubhashkumar at gmail.com  Wed Sep 12 19:36:38 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Wed, 12 Sep 2012 08:36:38 -0700
Subject: [Freeswitch-users] Bridge after Leg B has connected
In-Reply-To: <CAKzWOxUgXFMTEVH_6Zx55Ga2CSQNmQuEsLPOFpGBan85MM3Z4g@mail.gmail.com>
References: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
	<CAKzWOxUgXFMTEVH_6Zx55Ga2CSQNmQuEsLPOFpGBan85MM3Z4g@mail.gmail.com>
Message-ID: <CAFvodxU+0agXtS-NsH5EWRJhW5yecQE_XbgvgjVu6Kqf0um6qA@mail.gmail.com>

Hi,

Is there any variable to get the information about the call b leg status?

Because based on the call b leg hangup cause we are trying to do call retry
mechanism in our app.
On Sep 12, 2012 8:43 PM, "Michael Collins" <msc at freeswitch.org> wrote:

>
>
> On Wed, Sep 12, 2012 at 2:21 AM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>
>> Hi,
>>
>>  I have a simple dialplan that just do bridge for incoming calls. I'm
>> trying to do bridge the Call Leg A with Call Leg B after getting
>> confirmation of call connected from Call Leg B
>>
>>   <action application="bridge" data="(sofia or
>> freetdm)/internal/destination number"/> **
>>
>> Is there any way to bridge the both the call legs after getting
>> confirmation from call leg B?
>>
> The legs will be bridged as soon as the B leg returns media (SIP 180/183)
> or in the case of ignoring early media it bridges when receiving a SIP 200
> OK. In other words, the bridge app does all the work for you.
>
> Now, if you want the called party to press a key to acknowledge receipt of
> the call then you need the 'answer confirmation' feature:
>
> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> HtH,
> MC
>
>>
>> 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
>>
>>
>
>
> --
> 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 spencer at 5ninesolutions.com  Wed Sep 12 19:39:13 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 12 Sep 2012 08:39:13 -0700
Subject: [Freeswitch-users] Stuck Sessions
Message-ID: <23C91B62-010B-464A-A7A3-3EE996B5A2B9@5ninesolutions.com>

Hello,
Recently upgraded a few of our boxes to git head and I'm seeing the following behavior:

freeswitch at internal> status
UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539 microseconds
FreeSWITCH is ready
218 session(s) since startup
13 session(s) 0/200
5000 session(s) max
min idle cpu 0.00/99.00
Current Stack Size/Max 240K/240K

freeswitch at internal> show calls

0 total.

freeswitch at internal> show channels

0 total.

Note the discrepancy in the active sessions.  I'm happy to file a jira but I'm not sure of the necessary debugging info needed.  This particular machine in using a dialplan generated by FusionPBX and is close to stock.  The only scripting is a huntgroup which checks if session:ready().

Thanks,
Spencer

  

From krice at freeswitch.org  Wed Sep 12 19:39:45 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Sep 2012 10:39:45 -0500
Subject: [Freeswitch-users] Is the FS JIRA server not working properly
 now?
In-Reply-To: <50509A3D.7010500@hw.ac.uk>
Message-ID: <CC7616F1.4D038%krice@freeswitch.org>

Its possible...we just did an upgrade of Jira... I'll check it out...

K


On 9/12/12 9:20 AM, "x.liu" <x.liu at hw.ac.uk> wrote:

> Hi,
> 
> Normally after I add a comment to an existing issue I will receive an
> email copy of the comment very soon.
> 
> I haven't received the email after I added a comment 6 hours ago today.
> so I am wondering if there is something wrong.
> 
> Cheers,
> 
> Xing
> 
> 
> 
> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From krice at freeswitch.org  Wed Sep 12 19:43:06 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Sep 2012 10:43:06 -0500
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <23C91B62-010B-464A-A7A3-3EE996B5A2B9@5ninesolutions.com>
Message-ID: <CC7617BA.4D03C%krice@freeswitch.org>

Update... This issue was fixed already...


On 9/12/12 10:39 AM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:

> Hello,
> Recently upgraded a few of our boxes to git head and I'm seeing the following
> behavior:
> 
> freeswitch at internal> status
> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539
> microseconds
> FreeSWITCH is ready
> 218 session(s) since startup
> 13 session(s) 0/200
> 5000 session(s) max
> min idle cpu 0.00/99.00
> Current Stack Size/Max 240K/240K
> 
> freeswitch at internal> show calls
> 
> 0 total.
> 
> freeswitch at internal> show channels
> 
> 0 total.
> 
> Note the discrepancy in the active sessions.  I'm happy to file a jira but I'm
> not sure of the necessary debugging info needed.  This particular machine in
> using a dialplan generated by FusionPBX and is close to stock.  The only
> scripting is a huntgroup which checks if session:ready().
> 
> Thanks,
> Spencer
> 
>   
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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




From avi at avimarcus.net  Wed Sep 12 19:44:58 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 12 Sep 2012 18:44:58 +0300
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <23C91B62-010B-464A-A7A3-3EE996B5A2B9@5ninesolutions.com>
References: <23C91B62-010B-464A-A7A3-3EE996B5A2B9@5ninesolutions.com>
Message-ID: <CALNZuw=swJQ0tQSuXTo-y1Lvq8Hkj7PQHZ+f0i0+YZR-55xMGQ@mail.gmail.com>

How recent? I'm seeing this too, but on a build from 2 weeks ago:

FreeSWITCH Version 1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at
commit f1d201f03b on Tue, 28 Aug 2012 23:05:59 Z)

At one point FS stopped responding to calls until I did a "hupall" to hang
up all the stuck sessions. (I first tried to restart the profile, but that
didn't work until I did the hupall)

-Avi


On Wed, Sep 12, 2012 at 6:39 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello,
> Recently upgraded a few of our boxes to git head and I'm seeing the
> following behavior:
>
> freeswitch at internal> status
> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539
> microseconds
> FreeSWITCH is ready
> 218 session(s) since startup
> 13 session(s) 0/200
> 5000 session(s) max
> min idle cpu 0.00/99.00
> Current Stack Size/Max 240K/240K
>
> freeswitch at internal> show calls
>
> 0 total.
>
> freeswitch at internal> show channels
>
> 0 total.
>
> Note the discrepancy in the active sessions.  I'm happy to file a jira but
> I'm not sure of the necessary debugging info needed.  This particular
> machine in using a dialplan generated by FusionPBX and is close to stock.
>  The only scripting is a huntgroup which checks if session:ready().
>
> Thanks,
> Spencer
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 12 19:54:28 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 12 Sep 2012 08:54:28 -0700
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <CC7617BA.4D03C%krice@freeswitch.org>
References: <CC7617BA.4D03C%krice@freeswitch.org>
Message-ID: <66F762EC-8357-48D5-A5D6-73AF28C2A9B7@5ninesolutions.com>

Ken,
I'm running rev 8f0c726b136aa5518c4b52e76107031fec5a8756.  The only commit since this is:

commit 631c976f59ede25c3e9797100861bfc4b87b7424
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Sep 12 10:22:03 2012 -0500

    don't put casue from unanswered pickups as result of originate

Is there an open Jira that I can update?


On Sep 12, 2012, at 8:43 AM, Ken Rice wrote:

> Update... This issue was fixed already...
> 
> 
> On 9/12/12 10:39 AM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:
> 
>> Hello,
>> Recently upgraded a few of our boxes to git head and I'm seeing the following
>> behavior:
>> 
>> freeswitch at internal> status
>> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539
>> microseconds
>> FreeSWITCH is ready
>> 218 session(s) since startup
>> 13 session(s) 0/200
>> 5000 session(s) max
>> min idle cpu 0.00/99.00
>> Current Stack Size/Max 240K/240K
>> 
>> freeswitch at internal> show calls
>> 
>> 0 total.
>> 
>> freeswitch at internal> show channels
>> 
>> 0 total.
>> 
>> Note the discrepancy in the active sessions.  I'm happy to file a jira but I'm
>> not sure of the necessary debugging info needed.  This particular machine in
>> using a dialplan generated by FusionPBX and is close to stock.  The only
>> scripting is a huntgroup which checks if session:ready().
>> 
>> Thanks,
>> Spencer
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> 



From krice at freeswitch.org  Wed Sep 12 19:55:18 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Sep 2012 10:55:18 -0500
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <CALNZuw=swJQ0tQSuXTo-y1Lvq8Hkj7PQHZ+f0i0+YZR-55xMGQ@mail.gmail.com>
Message-ID: <CC761A96.4D03E%krice@freeswitch.org>

It was fixed about that long ago...


On 9/12/12 10:44 AM, "Avi Marcus" <avi at avimarcus.net> wrote:

> How recent? I'm seeing this too, but on a build from 2 weeks ago:
> 
> FreeSWITCH Version 1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at commit
> f1d201f03b on Tue, 28 Aug 2012 23:05:59 Z)
> 
> At one point FS stopped responding to calls until I did a "hupall" to hang up
> all the stuck sessions. (I first tried to restart the profile, but that didn't
> work until I did the hupall)
> 
> -Avi
> 
> 
> On Wed, Sep 12, 2012 at 6:39 PM, Spencer Thomason <spencer at 5ninesolutions.com>
> wrote:
>> Hello,
>> Recently upgraded a few of our boxes to git head and I'm seeing the following
>> behavior:
>> 
>> freeswitch at internal> status
>> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539
>> microseconds
>> FreeSWITCH is ready
>> 218 session(s) since startup
>> 13 session(s) 0/200
>> 5000 session(s) max
>> min idle cpu 0.00/99.00
>> Current Stack Size/Max 240K/240K
>> 
>> freeswitch at internal> show calls
>> 
>> 0 total.
>> 
>> freeswitch at internal> show channels
>> 
>> 0 total.
>> 
>> Note the discrepancy in the active sessions. ?I'm happy to file a jira but
>> I'm not sure of the necessary debugging info needed. ?This particular machine
>> in using a dialplan generated by FusionPBX and is close to stock. ?The only
>> scripting is a huntgroup which checks if session:ready().
>> 
>> Thanks,
>> Spencer
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 xyangni at gmail.com  Wed Sep 12 20:01:37 2012
From: xyangni at gmail.com (Yihui Li)
Date: Wed, 12 Sep 2012 17:01:37 +0100
Subject: [Freeswitch-users] GSMOPEN to send long SMS
In-Reply-To: <CAAyX+KHgZGmLWykAdfXg=ry62ossNRtZb3Rts70u=Byq3QY4ZQ@mail.gmail.com>
References: <CAO1DH1_1CW8SWpV1BrZUHByg9GsbbMfgeY7jp=ig8RbCUQws1A@mail.gmail.com>
	<CAAyX+KHgZGmLWykAdfXg=ry62ossNRtZb3Rts70u=Byq3QY4ZQ@mail.gmail.com>
Message-ID: <CAO1DH1_ObH0u9fnAb3Of5+XCmLyPCNGrJn6mnuCBHxu0B+njSA@mail.gmail.com>

If send it in this way, the parts will not be combined into 1 message on
the receiver's mobile.
The order of the 2 or more parts may also be revered, which will make it
difficult to read.

Regards,
Eric

On Wed, Sep 12, 2012 at 2:37 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Send the message in multiple parts of <= 160 chars each?
>
> Regards,
> Ben Langfeld
>
>
> On 12 September 2012 14:18, Yihui Li <xyangni at gmail.com> wrote:
>
>> Hi,
>>
>> I have used mod_gsmopen to send short text messages from my E1550.
>> However, where the length is longer than 160 chars, it always fail. Is
>> there any solution to send long sms like the mobile phone?
>>
>> Thanks,
>> Eric
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 12 20:01:51 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Sep 2012 11:01:51 -0500
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <66F762EC-8357-48D5-A5D6-73AF28C2A9B7@5ninesolutions.com>
Message-ID: <CC761C1F.4D043%krice@freeswitch.org>

That should have been fixed around

commit 3c685bff255779d21c1d6cd577276f1861be7aa2
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Aug 29 10:38:37 2012 -0500
    fix double sessions on auth challenged calls

On 9/12/12 10:54 AM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:

> Ken,
> I'm running rev 8f0c726b136aa5518c4b52e76107031fec5a8756.  The only commit
> since this is:
> 
> commit 631c976f59ede25c3e9797100861bfc4b87b7424
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Wed Sep 12 10:22:03 2012 -0500
> 
>     don't put casue from unanswered pickups as result of originate
> 
> Is there an open Jira that I can update?
> 
> 
> On Sep 12, 2012, at 8:43 AM, Ken Rice wrote:
> 
>> Update... This issue was fixed already...
>> 
>> 
>> On 9/12/12 10:39 AM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:
>> 
>>> Hello,
>>> Recently upgraded a few of our boxes to git head and I'm seeing the
>>> following
>>> behavior:
>>> 
>>> freeswitch at internal> status
>>> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539
>>> microseconds
>>> FreeSWITCH is ready
>>> 218 session(s) since startup
>>> 13 session(s) 0/200
>>> 5000 session(s) max
>>> min idle cpu 0.00/99.00
>>> Current Stack Size/Max 240K/240K
>>> 
>>> freeswitch at internal> show calls
>>> 
>>> 0 total.
>>> 
>>> freeswitch at internal> show channels
>>> 
>>> 0 total.
>>> 
>>> Note the discrepancy in the active sessions.  I'm happy to file a jira but
>>> I'm
>>> not sure of the necessary debugging info needed.  This particular machine in
>>> using a dialplan generated by FusionPBX and is close to stock.  The only
>>> scripting is a huntgroup which checks if session:ready().
>>> 
>>> Thanks,
>>> Spencer
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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




From andrew at cassidywebservices.co.uk  Wed Sep 12 20:33:47 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 12 Sep 2012 17:33:47 +0100
Subject: [Freeswitch-users] GSMOPEN to send long SMS
In-Reply-To: <CAO1DH1_ObH0u9fnAb3Of5+XCmLyPCNGrJn6mnuCBHxu0B+njSA@mail.gmail.com>
References: <CAO1DH1_1CW8SWpV1BrZUHByg9GsbbMfgeY7jp=ig8RbCUQws1A@mail.gmail.com>
	<CAAyX+KHgZGmLWykAdfXg=ry62ossNRtZb3Rts70u=Byq3QY4ZQ@mail.gmail.com>
	<CAO1DH1_ObH0u9fnAb3Of5+XCmLyPCNGrJn6mnuCBHxu0B+njSA@mail.gmail.com>
Message-ID: <CAMbW9o+K1qhSbfxoTK=kLR==E3g7wbPxuTq07u-+fCoDMu6OVQ@mail.gmail.com>

http://en.wikipedia.org/wiki/Concatenated_SMS

Just some random info. Obviosuly will need a test.

On 12 September 2012 17:01, Yihui Li <xyangni at gmail.com> wrote:

> If send it in this way, the parts will not be combined into 1 message on
> the receiver's mobile.
> The order of the 2 or more parts may also be revered, which will make it
> difficult to read.
>
> Regards,
> Eric
>
>
> On Wed, Sep 12, 2012 at 2:37 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Send the message in multiple parts of <= 160 chars each?
>>
>> Regards,
>> Ben Langfeld
>>
>>
>> On 12 September 2012 14:18, Yihui Li <xyangni at gmail.com> wrote:
>>
>>>  Hi,
>>>
>>> I have used mod_gsmopen to send short text messages from my E1550.
>>> However, where the length is longer than 160 chars, it always fail. Is
>>> there any solution to send long sms like the mobile phone?
>>>
>>> Thanks,
>>> Eric
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 djbinter at gmail.com  Wed Sep 12 21:29:56 2012
From: djbinter at gmail.com (DJB International)
Date: Wed, 12 Sep 2012 10:29:56 -0700
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <CC761C1F.4D043%krice@freeswitch.org>
References: <66F762EC-8357-48D5-A5D6-73AF28C2A9B7@5ninesolutions.com>
	<CC761C1F.4D043%krice@freeswitch.org>
Message-ID: <CAAB3jWt16OwExaZguTEu2-sRsp2YSd3KotjfC+LQi-gqq0_DAQ@mail.gmail.com>

I had this issue as well and opened jira ticket:
http://jira.freeswitch.org/browse/FS-4598, but it should be fixed after git
at commit ff3631b474 on Tue, 04 Sep 2012 16:13:52 Z, but I did not have a
chance to update and test yet.

-djbinter


On Wed, Sep 12, 2012 at 9:01 AM, Ken Rice <krice at freeswitch.org> wrote:

> That should have been fixed around
>
> commit 3c685bff255779d21c1d6cd577276f1861be7aa2
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Wed Aug 29 10:38:37 2012 -0500
>     fix double sessions on auth challenged calls
>
> On 9/12/12 10:54 AM, "Spencer Thomason" <spencer at 5ninesolutions.com>
> wrote:
>
> > Ken,
> > I'm running rev 8f0c726b136aa5518c4b52e76107031fec5a8756.  The only
> commit
> > since this is:
> >
> > commit 631c976f59ede25c3e9797100861bfc4b87b7424
> > Author: Anthony Minessale <anthm at freeswitch.org>
> > Date:   Wed Sep 12 10:22:03 2012 -0500
> >
> >     don't put casue from unanswered pickups as result of originate
> >
> > Is there an open Jira that I can update?
> >
> >
> > On Sep 12, 2012, at 8:43 AM, Ken Rice wrote:
> >
> >> Update... This issue was fixed already...
> >>
> >>
> >> On 9/12/12 10:39 AM, "Spencer Thomason" <spencer at 5ninesolutions.com>
> wrote:
> >>
> >>> Hello,
> >>> Recently upgraded a few of our boxes to git head and I'm seeing the
> >>> following
> >>> behavior:
> >>>
> >>> freeswitch at internal> status
> >>> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds,
> 539
> >>> microseconds
> >>> FreeSWITCH is ready
> >>> 218 session(s) since startup
> >>> 13 session(s) 0/200
> >>> 5000 session(s) max
> >>> min idle cpu 0.00/99.00
> >>> Current Stack Size/Max 240K/240K
> >>>
> >>> freeswitch at internal> show calls
> >>>
> >>> 0 total.
> >>>
> >>> freeswitch at internal> show channels
> >>>
> >>> 0 total.
> >>>
> >>> Note the discrepancy in the active sessions.  I'm happy to file a jira
> but
> >>> I'm
> >>> not sure of the necessary debugging info needed.  This particular
> machine in
> >>> using a dialplan generated by FusionPBX and is close to stock.  The
> only
> >>> scripting is a huntgroup which checks if session:ready().
> >>>
> >>> Thanks,
> >>> Spencer
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
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From spencer at 5ninesolutions.com  Wed Sep 12 21:36:12 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 12 Sep 2012 10:36:12 -0700
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <CAAB3jWt16OwExaZguTEu2-sRsp2YSd3KotjfC+LQi-gqq0_DAQ@mail.gmail.com>
References: <66F762EC-8357-48D5-A5D6-73AF28C2A9B7@5ninesolutions.com>
	<CC761C1F.4D043%krice@freeswitch.org>
	<CAAB3jWt16OwExaZguTEu2-sRsp2YSd3KotjfC+LQi-gqq0_DAQ@mail.gmail.com>
Message-ID: <A0EDD5AD-25C0-4F3A-90E6-0F57303E7E43@5ninesolutions.com>

I'll update this ticket.

Thanks,
Spencer



On Sep 12, 2012, at 10:29 AM, DJB International wrote:

> I had this issue as well and opened jira ticket:  http://jira.freeswitch.org/browse/FS-4598, but it should be fixed after git at commit ff3631b474 on Tue, 04 Sep 2012 16:13:52 Z, but I did not have a chance to update and test yet.
> 
> -djbinter
> 
> 
> On Wed, Sep 12, 2012 at 9:01 AM, Ken Rice <krice at freeswitch.org> wrote:
> That should have been fixed around
> 
> commit 3c685bff255779d21c1d6cd577276f1861be7aa2
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Wed Aug 29 10:38:37 2012 -0500
>     fix double sessions on auth challenged calls
> 
> On 9/12/12 10:54 AM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:
> 
> > Ken,
> > I'm running rev 8f0c726b136aa5518c4b52e76107031fec5a8756.  The only commit
> > since this is:
> >
> > commit 631c976f59ede25c3e9797100861bfc4b87b7424
> > Author: Anthony Minessale <anthm at freeswitch.org>
> > Date:   Wed Sep 12 10:22:03 2012 -0500
> >
> >     don't put casue from unanswered pickups as result of originate
> >
> > Is there an open Jira that I can update?
> >
> >
> > On Sep 12, 2012, at 8:43 AM, Ken Rice wrote:
> >
> >> Update... This issue was fixed already...
> >>
> >>
> >> On 9/12/12 10:39 AM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:
> >>
> >>> Hello,
> >>> Recently upgraded a few of our boxes to git head and I'm seeing the
> >>> following
> >>> behavior:
> >>>
> >>> freeswitch at internal> status
> >>> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539
> >>> microseconds
> >>> FreeSWITCH is ready
> >>> 218 session(s) since startup
> >>> 13 session(s) 0/200
> >>> 5000 session(s) max
> >>> min idle cpu 0.00/99.00
> >>> Current Stack Size/Max 240K/240K
> >>>
> >>> freeswitch at internal> show calls
> >>>
> >>> 0 total.
> >>>
> >>> freeswitch at internal> show channels
> >>>
> >>> 0 total.
> >>>
> >>> Note the discrepancy in the active sessions.  I'm happy to file a jira but
> >>> I'm
> >>> not sure of the necessary debugging info needed.  This particular machine in
> >>> using a dialplan generated by FusionPBX and is close to stock.  The only
> >>> scripting is a huntgroup which checks if session:ready().
> >>>
> >>> Thanks,
> >>> Spencer
> >>>
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Wed Sep 12 22:23:53 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Sep 2012 13:23:53 -0500
Subject: [Freeswitch-users] Stuck Sessions
In-Reply-To: <A0EDD5AD-25C0-4F3A-90E6-0F57303E7E43@5ninesolutions.com>
References: <66F762EC-8357-48D5-A5D6-73AF28C2A9B7@5ninesolutions.com>
	<CC761C1F.4D043%krice@freeswitch.org>
	<CAAB3jWt16OwExaZguTEu2-sRsp2YSd3KotjfC+LQi-gqq0_DAQ@mail.gmail.com>
	<A0EDD5AD-25C0-4F3A-90E6-0F57303E7E43@5ninesolutions.com>
Message-ID: <CAKbxfG9PDQdDN4hEdNf96xzuEMTGvAbPmx43=0gsA0t7Kq+1Vg@mail.gmail.com>

We always test the fixes AFTER the releases instead of before.
I guess we need fake releases again.....


On Wed, Sep 12, 2012 at 12:36 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I'll update this ticket.
>
> Thanks,
> Spencer
>
>
>
> On Sep 12, 2012, at 10:29 AM, DJB International wrote:
>
> I had this issue as well and opened jira ticket:
> http://jira.freeswitch.org/browse/FS-4598, but it should be fixed after
> git at commit ff3631b474 on Tue, 04 Sep 2012 16:13:52 Z, but I did not have
> a chance to update and test yet.
>
> -djbinter
>
>
> On Wed, Sep 12, 2012 at 9:01 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>> That should have been fixed around
>>
>> commit 3c685bff255779d21c1d6cd577276f1861be7aa2
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date:   Wed Aug 29 10:38:37 2012 -0500
>>     fix double sessions on auth challenged calls
>>
>> On 9/12/12 10:54 AM, "Spencer Thomason" <spencer at 5ninesolutions.com>
>> wrote:
>>
>> > Ken,
>> > I'm running rev 8f0c726b136aa5518c4b52e76107031fec5a8756.  The only
>> commit
>> > since this is:
>> >
>> > commit 631c976f59ede25c3e9797100861bfc4b87b7424
>> > Author: Anthony Minessale <anthm at freeswitch.org>
>> > Date:   Wed Sep 12 10:22:03 2012 -0500
>> >
>> >     don't put casue from unanswered pickups as result of originate
>> >
>> > Is there an open Jira that I can update?
>> >
>> >
>> > On Sep 12, 2012, at 8:43 AM, Ken Rice wrote:
>> >
>> >> Update... This issue was fixed already...
>> >>
>> >>
>> >> On 9/12/12 10:39 AM, "Spencer Thomason" <spencer at 5ninesolutions.com>
>> wrote:
>> >>
>> >>> Hello,
>> >>> Recently upgraded a few of our boxes to git head and I'm seeing the
>> >>> following
>> >>> behavior:
>> >>>
>> >>> freeswitch at internal> status
>> >>> UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552
>> milliseconds, 539
>> >>> microseconds
>> >>> FreeSWITCH is ready
>> >>> 218 session(s) since startup
>> >>> 13 session(s) 0/200
>> >>> 5000 session(s) max
>> >>> min idle cpu 0.00/99.00
>> >>> Current Stack Size/Max 240K/240K
>> >>>
>> >>> freeswitch at internal> show calls
>> >>>
>> >>> 0 total.
>> >>>
>> >>> freeswitch at internal> show channels
>> >>>
>> >>> 0 total.
>> >>>
>> >>> Note the discrepancy in the active sessions.  I'm happy to file a
>> jira but
>> >>> I'm
>> >>> not sure of the necessary debugging info needed.  This particular
>> machine in
>> >>> using a dialplan generated by FusionPBX and is close to stock.  The
>> only
>> >>> scripting is a huntgroup which checks if session:ready().
>> >>>
>> >>> Thanks,
>> >>> Spencer
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at 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
>> >>
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 phil.kim at valiant.com  Thu Sep 13 01:16:11 2012
From: phil.kim at valiant.com (Phil Kim)
Date: Wed, 12 Sep 2012 17:16:11 -0400
Subject: [Freeswitch-users] Static sound on outbound calls
Message-ID: <C69FF59885AC464896A4B68500B6E97409BFA479@VAL-SV-EXCH-01.valiant.com>

Hello All,

 

I have been using FreeSWITCH over the last two years on both inbound and
outbound dialing.

 

I have a problem with Outbound calling.

 

This does not occur all the time, but often enough for us to be
concerned.

 

We have an IVR system that make an outbound call to our client and
process the call flow using DTMF.

 

Often times when user answers the call, there is nothing but static
sound. It does not respond to any DTMF, and does not play any sound file
either. 

 

It just hangs there with little static sound in background. 

 

I have consulted my SIP provider but they have no idea either.

 

Any one has any idea what might be causing the issue?

 

Thank you in advance.

 

 

Best regards,

 

Phil H. Kim

Software Engineer

 

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 msc at freeswitch.org  Thu Sep 13 02:02:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Sep 2012 15:02:58 -0700
Subject: [Freeswitch-users] Static sound on outbound calls
In-Reply-To: <C69FF59885AC464896A4B68500B6E97409BFA479@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E97409BFA479@VAL-SV-EXCH-01.valiant.com>
Message-ID: <CAKzWOxWzqp4-DvcStJw5HSYSB1VAYG6Jrp6zrSfeyCY7=2hyNQ@mail.gmail.com>

Hi Phil,

You are at the point where you need to catch the culprit in the act. My
favorite tool for this is a packet capture. I recommend you check out the
tools listed here:
http://wiki.freeswitch.org/wiki/Packet_Capture

pcapsipdump is one of my favorites because it picks out the voip calls and
puts them in separate files. However, in your case you may want to consider
using tcpdump with a ring buffer and capture all the traffic on the NIC.
The reason I say that is because when it comes time to do the analysis you
may see other network traffic that coincides with the static you are
hearing.

-MC

On Wed, Sep 12, 2012 at 2:16 PM, Phil Kim <phil.kim at valiant.com> wrote:

>  Hello All,****
>
> ** **
>
> I have been using FreeSWITCH over the last two years on both inbound and
> outbound dialing.****
>
> ** **
>
> I have a problem with Outbound calling.****
>
> ** **
>
> This does not occur all the time, but often enough for us to be concerned.
> ****
>
> ** **
>
> We have an IVR system that make an outbound call to our client and process
> the call flow using DTMF.****
>
> ** **
>
> Often times when user answers the call, there is nothing but static sound.
> It does not respond to any DTMF, and does not play any sound file either.
> ****
>
> ** **
>
> It just hangs there with little static sound in background. ****
>
> ** **
>
> I have consulted my SIP provider but they have no idea either.****
>
> ** **
>
> Any one has any idea what might be causing the issue?****
>
> ** **
>
> Thank you in advance.****
>
> ** **
>
> ** **
>
> Best regards,****
>
> ** **
>
> Phil H. Kim****
>
> Software Engineer****
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lists at kavun.ch  Thu Sep 13 10:34:00 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Sep 2012 02:34:00 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
Message-ID: <34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>

Thanks Michael. I will check out the stable version and let you know.

Regards,
Emrah
On Sep 11, 2012, at 8:02 PM, Michael Collins <msc at freeswitch.org> wrote:

> I do not see this behavior on 1.2.stable branch. When I press a key during the invalid prompt it breaks out of playback immediately and attempts to process whatever digit(s) I input. I tested with x5000 sample IVR:
> Dial 5000
> Press 7, wait for "invalid entry" message to start playing
> Press 3 while she's saying "that was an invalid entry"
> After a few seconds MOH comes on.
> 
> -MC
> 
> On Tue, Sep 11, 2012 at 12:43 PM, Emrah <lists at kavun.ch> wrote:
> Just wanted to drop a quick note about an issue I just stumbled upon.
> 
> When using play_and_get_digits, you are still able to input DTMFs when the invalid-entry prompt is played, but the message won't be interrupted and your DTMF input won't be executed until the prompt is fully played.
> 
> Can someone look into this and confirm?
> 
> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Sep 13 11:08:03 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Sep 2012 03:08:03 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
Message-ID: <D2C38CA4-ABDC-4DEC-8C91-E2725A612A00@kavun.ch>

Hey Michael,

Just tested the demo IVR. It works indeed, but it doesn't use the play_and_get_digits app.

Anything else I should try?

Best to you and thanks for your input,
Emrah

On Sep 11, 2012, at 8:02 PM, Michael Collins <msc at freeswitch.org> wrote:

> I do not see this behavior on 1.2.stable branch. When I press a key during the invalid prompt it breaks out of playback immediately and attempts to process whatever digit(s) I input. I tested with x5000 sample IVR:
> Dial 5000
> Press 7, wait for "invalid entry" message to start playing
> Press 3 while she's saying "that was an invalid entry"
> After a few seconds MOH comes on.
> 
> -MC
> 
> On Tue, Sep 11, 2012 at 12:43 PM, Emrah <lists at kavun.ch> wrote:
> Just wanted to drop a quick note about an issue I just stumbled upon.
> 
> When using play_and_get_digits, you are still able to input DTMFs when the invalid-entry prompt is played, but the message won't be interrupted and your DTMF input won't be executed until the prompt is fully played.
> 
> Can someone look into this and confirm?
> 
> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Sep 13 11:26:31 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Sep 2012 03:26:31 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
Message-ID: <7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>

This happens with the stable version as well. It seems to be limited to the play_and_get_digits app though.

Best,
Emrah
On Sep 13, 2012, at 2:34 AM, Emrah <lists at kavun.ch> wrote:

> Thanks Michael. I will check out the stable version and let you know.
> 
> Regards,
> Emrah
> On Sep 11, 2012, at 8:02 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> I do not see this behavior on 1.2.stable branch. When I press a key during the invalid prompt it breaks out of playback immediately and attempts to process whatever digit(s) I input. I tested with x5000 sample IVR:
>> Dial 5000
>> Press 7, wait for "invalid entry" message to start playing
>> Press 3 while she's saying "that was an invalid entry"
>> After a few seconds MOH comes on.
>> 
>> -MC
>> 
>> On Tue, Sep 11, 2012 at 12:43 PM, Emrah <lists at kavun.ch> wrote:
>> Just wanted to drop a quick note about an issue I just stumbled upon.
>> 
>> When using play_and_get_digits, you are still able to input DTMFs when the invalid-entry prompt is played, but the message won't be interrupted and your DTMF input won't be executed until the prompt is fully played.
>> 
>> Can someone look into this and confirm?
>> 
>> 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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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  Thu Sep 13 12:36:53 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 13 Sep 2012 04:36:53 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
Message-ID: <27905.1347525413@ccs.covici.com>

I am not seeing this -- I can hit the dtmf while the prompt is still
playing.  When its playing the invalid prompt, what could you type
anyway -- it may not be listening yet.

Emrah <lists at kavun.ch> wrote:

> This happens with the stable version as well. It seems to be limited to the play_and_get_digits app though.
> 
> Best,
> Emrah
> On Sep 13, 2012, at 2:34 AM, Emrah <lists at kavun.ch> wrote:
> 
> > Thanks Michael. I will check out the stable version and let you know.
> > 
> > Regards,
> > Emrah
> > On Sep 11, 2012, at 8:02 PM, Michael Collins <msc at freeswitch.org> wrote:
> > 
> >> I do not see this behavior on 1.2.stable branch. When I press a key during the invalid prompt it breaks out of playback immediately and attempts to process whatever digit(s) I input. I tested with x5000 sample IVR:
> >> Dial 5000
> >> Press 7, wait for "invalid entry" message to start playing
> >> Press 3 while she's saying "that was an invalid entry"
> >> After a few seconds MOH comes on.
> >> 
> >> -MC
> >> 
> >> On Tue, Sep 11, 2012 at 12:43 PM, Emrah <lists at kavun.ch> wrote:
> >> Just wanted to drop a quick note about an issue I just stumbled upon.
> >> 
> >> When using play_and_get_digits, you are still able to input DTMFs when the invalid-entry prompt is played, but the message won't be interrupted and your DTMF input won't be executed until the prompt is fully played.
> >> 
> >> Can someone look into this and confirm?
> >> 
> >> 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
> >> 
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> 
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

         John Covici
         covici at ccs.covici.com


From david.villasmil.work at gmail.com  Thu Sep 13 12:55:11 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 13 Sep 2012 10:55:11 +0200
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <1347457994946-7582782.post@n2.nabble.com>
References: <4F87EF60.3070105@vts24.ru>
	<CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
	<4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru>
	<CAKzWOxVs4qWH2ECOORXDQob5AOCrznitJabvLj+N+GGZ4Dzjig@mail.gmail.com>
	<4FAB719E.9010800@vts24.ru> <4FB4C69A.7070108@vts24.ru>
	<CANBhyPB2dZuAqG3CraDP=Hv8VU-c_MbF8AMOuruVpicQh_5YyQ@mail.gmail.com>
	<1347457994946-7582782.post@n2.nabble.com>
Message-ID: <CAFGRPVofDWHgZXU-wOTP-rcN=OZLMMe1zy+oqU6Znz28k3NQ7g@mail.gmail.com>

Hello guys,

the way you have it:

<action application="set" data="continue_on_fail=1,2,3,
6,25,34,38,41,42,44,47,63,66,500,501"/>

Never worked for me, I had to set it up like:

<action application="set" data="
continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER"/>


With the description text, not the error number...

Hope this helps you..


David

On Wed, Sep 12, 2012 at 3:53 PM, iam <iam at onnet.su> wrote:

> Hi Vitaly!
>
> Did you ever solve this problem?
> I have the same continue_on_fail behaviour.
>
> http://jira.freeswitch.org/browse/FS-4232
>
> Vitaly added a comment - 26/Jun/12 3:14 AM
> Sorry for the long absence, but the problem is still actual for me.
> I replaced in my dialplan continue_on_fail to "true", but it's not give
> desired effect.
> May be some additional variables can help me?
>
> Regards,
> Kirill
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.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
>
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From iam at onnet.su  Thu Sep 13 14:14:47 2012
From: iam at onnet.su (Kirill Sysoev)
Date: Thu, 13 Sep 2012 14:14:47 +0400
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <CAFGRPVofDWHgZXU-wOTP-rcN=OZLMMe1zy+oqU6Znz28k3NQ7g@mail.gmail.com>
References: <4F87EF60.3070105@vts24.ru>
	<CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
	<4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru>
	<CAKzWOxVs4qWH2ECOORXDQob5AOCrznitJabvLj+N+GGZ4Dzjig@mail.gmail.com>
	<4FAB719E.9010800@vts24.ru> <4FB4C69A.7070108@vts24.ru>
	<CANBhyPB2dZuAqG3CraDP=Hv8VU-c_MbF8AMOuruVpicQh_5YyQ@mail.gmail.com>
	<1347457994946-7582782.post@n2.nabble.com>
	<CAFGRPVofDWHgZXU-wOTP-rcN=OZLMMe1zy+oqU6Znz28k3NQ7g@mail.gmail.com>
Message-ID: <5051B217.3090300@onnet.su>

Hi David!

Thank you for your answer!

It doesn't work either.

Here is the situation:
1. the phone call goes to cell phone through the first gateway.
2. cell phone ringing.
3. phone call rejected by cell phone ("Cancel" button pressed)
4. FS receives USER_BUSY from the first gateway
5. FS executes continue_on_fail despite there is no USER_BUSY in 
continue_on_fail cause list settings.

Brian suggested workaround in Vitaly's report ( 
http://jira.freeswitch.org/browse/FS-4232 )
But if the cell phone is off, caller can not hear announce about that.

So I was wondering whether it is solved in some another way.

DialPlan:

<condition field="destination_number" expression="^(8921\d{7}|8931\d{7})$">
     <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" 
data="continue_on_fail=CALL_REJECTED,NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION"/>
         <action application="bridge" data="sofia/gateway/gw1/$1"/>
         <action application="bridge" data="sofia/gateway/gw2/$1"/>
</condition>


Short callLogs:

2012-09-13 13:46:09.332094 [DEBUG] switch_channel.c:3057 
(sofia/onnet/78120000000 at y.y.y.y) Callstate Change RINGING -> EARLY
2012-09-13 13:46:09.332094 [DEBUG] mod_sofia.c:2643 Ring SDP:
v=0
o=FreeSWITCH 1347502619 1347502620 IN IP4 x.x.x.x
s=FreeSWITCH
c=IN IP4 x.x.x.x
t=0 0
m=audio 26950 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

2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:778 Send signal 
sofia/onnet/78120000000 at y.y.y.y [BREAK]
2012-09-13 13:46:09.332094 [DEBUG] switch_ivr_originate.c:3360 Originate 
Resulted in Success: [sofia/gw1/89210000000]
2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:778 Send signal 
sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:778 Send signal 
sofia/onnet/78120000000 at y.y.y.y [BREAK]
2012-09-13 13:46:09.332094 [DEBUG] switch_ivr_bridge.c:1359 
(sofia/gw1/89210000000) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:09.332094 [DEBUG] switch_core_state_machine.c:385 
(sofia/gw1/89210000000) Running State Change CS_EXCHANGE_MEDIA
2012-09-13 13:46:09.332094 [DEBUG] switch_core_state_machine.c:443 
(sofia/gw1/89210000000) State EXCHANGE_MEDIA
2012-09-13 13:46:09.332094 [DEBUG] mod_sofia.c:652 SOFIA EXCHANGE_MEDIA
2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:924 Send signal 
sofia/onnet/78120000000 at y.y.y.y [BREAK]
2012-09-13 13:46:09.332094 [DEBUG] sofia.c:6051 Channel 
sofia/onnet/78120000000 at y.y.y.y skipping state [early][183]
2012-09-13 13:46:09.372176 [DEBUG] switch_rtp.c:3594 Correct ip/port 
confirmed.
2012-09-13 13:46:09.432175 [DEBUG] switch_rtp.c:3594 Correct ip/port 
confirmed.
2012-09-13 13:46:13.632367 [DEBUG] switch_core_session.c:924 Send signal 
sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:13.632367 [DEBUG] switch_core_session.c:924 Send signal 
sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:13.632367 [DEBUG] switch_core_session.c:924 Send signal 
sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:13.652424 [DEBUG] sofia.c:6058 Channel 
sofia/gw1/89210000000 entering state [terminated][486]
2012-09-13 13:46:13.652424 [DEBUG] switch_channel.c:2914 
(sofia/gw1/89210000000) Callstate Change EARLY -> HANGUP
2012-09-13 13:46:13.652424 [NOTICE] sofia.c:6850 Hangup 
sofia/gw1/89210000000 [CS_EXCHANGE_MEDIA] [USER_BUSY]
2012-09-13 13:46:13.652424 [DEBUG] switch_channel.c:2937 Send signal 
sofia/gw1/89210000000 [KILL]
2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:13.652424 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD 
DONE [sofia/gw1/89210000000]
2012-09-13 13:46:13.652424 [DEBUG] switch_ivr_bridge.c:613 Send signal 
sofia/onnet/78120000000 at y.y.y.y [BREAK]
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:443 
(sofia/gw1/89210000000) State EXCHANGE_MEDIA going to sleep
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:385 
(sofia/gw1/89210000000) Running State Change CS_HANGUP
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:625 
(sofia/gw1/89210000000) State HANGUP
2012-09-13 13:46:13.652424 [DEBUG] mod_sofia.c:474 Channel 
sofia/gw1/89210000000 hanging up, cause: USER_BUSY
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:47 
sofia/gw1/89210000000 Standard HANGUP, cause: USER_BUSY
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:625 
(sofia/gw1/89210000000) State HANGUP going to sleep
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:416 
(sofia/gw1/89210000000) State Change CS_HANGUP -> CS_REPORTING
2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:385 
(sofia/gw1/89210000000) Running State Change CS_REPORTING
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:685 
(sofia/gw1/89210000000) State REPORTING
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:79 
sofia/gw1/89210000000 Standard REPORTING, cause: USER_BUSY
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:685 
(sofia/gw1/89210000000) State REPORTING going to sleep
2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:410 
(sofia/gw1/89210000000) State Change CS_REPORTING -> CS_DESTROY
2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1429 Session 
779 (sofia/gw1/89210000000) Locked, Waiting on external entities
2012-09-13 13:46:13.672425 [DEBUG] switch_ivr_bridge.c:501 
sofia/gw1/89210000000 ending bridge by request from write function
2012-09-13 13:46:13.672425 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD 
DONE [sofia/onnet/78120000000 at y.y.y.y]
2012-09-13 13:46:13.672425 [DEBUG] switch_ivr_bridge.c:613 Send signal 
sofia/gw1/89210000000 [BREAK]
2012-09-13 13:46:13.672425 [DEBUG] switch_core_session.c:778 Send signal 
sofia/onnet/78120000000 at y.y.y.y [BREAK]
EXECUTE sofia/onnet/78120000000 at y.y.y.y 
bridge(sofia/gateway/gw2/89210000000)


Regards,
Kirill


13.09.2012 12:55, David Villasmil ?????:
> Hello guys,
>
> the way you have it:
>
> <action application="set" 
> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/>
>
> Never worked for me, I had to set it up like:
>
> <action application="set" 
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER"/> 
>
>
> With the description text, not the error number...
>
> Hope this helps you..
>
>
> David
>
> On Wed, Sep 12, 2012 at 3:53 PM, iam <iam at onnet.su 
> <mailto:iam at onnet.su>> wrote:
>
>     Hi Vitaly!
>
>     Did you ever solve this problem?
>     I have the same continue_on_fail behaviour.
>
>     http://jira.freeswitch.org/browse/FS-4232
>
>     Vitaly added a comment - 26/Jun/12 3:14 AM
>     Sorry for the long absence, but the problem is still actual for me.
>     I replaced in my dialplan continue_on_fail to "true", but it's not
>     give
>     desired effect.
>     May be some additional variables can help me?
>
>     Regards,
>     Kirill
>
>
>
>     --
>     View this message in context:
>     http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.html
>     Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>     _________________________________________________________________________
>     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 david.villasmil.work at gmail.com  Thu Sep 13 14:30:55 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 13 Sep 2012 12:30:55 +0200
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <5051B217.3090300@onnet.su>
References: <4F87EF60.3070105@vts24.ru>
	<CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
	<4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru>
	<CAKzWOxVs4qWH2ECOORXDQob5AOCrznitJabvLj+N+GGZ4Dzjig@mail.gmail.com>
	<4FAB719E.9010800@vts24.ru> <4FB4C69A.7070108@vts24.ru>
	<CANBhyPB2dZuAqG3CraDP=Hv8VU-c_MbF8AMOuruVpicQh_5YyQ@mail.gmail.com>
	<1347457994946-7582782.post@n2.nabble.com>
	<CAFGRPVofDWHgZXU-wOTP-rcN=OZLMMe1zy+oqU6Znz28k3NQ7g@mail.gmail.com>
	<5051B217.3090300@onnet.su>
Message-ID: <CAFGRPVqx3u+UoYmgnT4TPU=K+CchbsxHTPOCYoM2OnR2D0AVNw@mail.gmail.com>

Hello Kirill,

Can you set the debug level?

"fsctl loglevel debug"

Also:

2012-09-13 13:46:09.332094 [DEBUG] sofia.c:6051 Channel
sofia/onnet/78120000000 at y.y.y.y skipping state [early][183]

Might be preventing you from getting the annoucement, try:

bridge_early_media=true

and remove de hangup

hangup_after_bridge=true

And try again..

I think FS considers receiving early media as a successful bridge... not
sure though.




David

On Thu, Sep 13, 2012 at 12:14 PM, Kirill Sysoev <iam at onnet.su> wrote:

>  Hi David!
>
> Thank you for your answer!
>
> It doesn't work either.
>
> Here is the situation:
> 1. the phone call goes to cell phone through the first gateway.
> 2. cell phone ringing.
> 3. phone call rejected by cell phone ("Cancel" button pressed)
> 4. FS receives USER_BUSY from the first gateway
> 5. FS executes continue_on_fail despite there is no USER_BUSY in
> continue_on_fail cause list settings.
>
> Brian suggested workaround in Vitaly's report (
> http://jira.freeswitch.org/browse/FS-4232 )
> But if the cell phone is off, caller can not hear announce about that.
>
> So I was wondering whether it is solved in some another way.
>
> DialPlan:
>
> <condition field="destination_number" expression="^(8921\d{7}|8931\d{7})$">
>
>     <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set"
> data="continue_on_fail=CALL_REJECTED,NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION"/>
>         <action application="bridge" data="sofia/gateway/gw1/$1"/>
>         <action application="bridge" data="sofia/gateway/gw2/$1"/>
> </condition>
>
>
> Short callLogs:
>
> 2012-09-13 13:46:09.332094 [DEBUG] switch_channel.c:3057 (
> sofia/onnet/78120000000 at y.y.y.y) Callstate Change RINGING -> EARLY
> 2012-09-13 13:46:09.332094 [DEBUG] mod_sofia.c:2643 Ring SDP:
> v=0
> o=FreeSWITCH 1347502619 1347502620 IN IP4 x.x.x.x
> s=FreeSWITCH
> c=IN IP4 x.x.x.x
> t=0 0
> m=audio 26950 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
>
> 2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:778 Send signal
> sofia/onnet/78120000000 at y.y.y.y [BREAK]
> 2012-09-13 13:46:09.332094 [DEBUG] switch_ivr_originate.c:3360 Originate
> Resulted in Success: [sofia/gw1/89210000000]
> 2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:778 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:778 Send signal
> sofia/onnet/78120000000 at y.y.y.y [BREAK]
> 2012-09-13 13:46:09.332094 [DEBUG] switch_ivr_bridge.c:1359
> (sofia/gw1/89210000000) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:1229 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:09.332094 [DEBUG] switch_core_state_machine.c:385
> (sofia/gw1/89210000000) Running State Change CS_EXCHANGE_MEDIA
> 2012-09-13 13:46:09.332094 [DEBUG] switch_core_state_machine.c:443
> (sofia/gw1/89210000000) State EXCHANGE_MEDIA
> 2012-09-13 13:46:09.332094 [DEBUG] mod_sofia.c:652 SOFIA EXCHANGE_MEDIA
> 2012-09-13 13:46:09.332094 [DEBUG] switch_core_session.c:924 Send signal
> sofia/onnet/78120000000 at y.y.y.y [BREAK]
> 2012-09-13 13:46:09.332094 [DEBUG] sofia.c:6051 Channel
> sofia/onnet/78120000000 at y.y.y.y skipping state [early][183]
> 2012-09-13 13:46:09.372176 [DEBUG] switch_rtp.c:3594 Correct ip/port
> confirmed.
> 2012-09-13 13:46:09.432175 [DEBUG] switch_rtp.c:3594 Correct ip/port
> confirmed.
> 2012-09-13 13:46:13.632367 [DEBUG] switch_core_session.c:924 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:13.632367 [DEBUG] switch_core_session.c:924 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:13.632367 [DEBUG] switch_core_session.c:924 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:13.652424 [DEBUG] sofia.c:6058 Channel
> sofia/gw1/89210000000 entering state [terminated][486]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_channel.c:2914
> (sofia/gw1/89210000000) Callstate Change EARLY -> HANGUP
> 2012-09-13 13:46:13.652424 [NOTICE] sofia.c:6850 Hangup
> sofia/gw1/89210000000 [CS_EXCHANGE_MEDIA] [USER_BUSY]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_channel.c:2937 Send signal
> sofia/gw1/89210000000 [KILL]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1229 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD
> DONE [sofia/gw1/89210000000]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_ivr_bridge.c:613 Send signal
> sofia/onnet/78120000000 at y.y.y.y [BREAK]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:443
> (sofia/gw1/89210000000) State EXCHANGE_MEDIA going to sleep
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:385
> (sofia/gw1/89210000000) Running State Change CS_HANGUP
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:625
> (sofia/gw1/89210000000) State HANGUP
> 2012-09-13 13:46:13.652424 [DEBUG] mod_sofia.c:474 Channel
> sofia/gw1/89210000000 hanging up, cause: USER_BUSY
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:47
> sofia/gw1/89210000000 Standard HANGUP, cause: USER_BUSY
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:625
> (sofia/gw1/89210000000) State HANGUP going to sleep
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:416
> (sofia/gw1/89210000000) State Change CS_HANGUP -> CS_REPORTING
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1229 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:385
> (sofia/gw1/89210000000) Running State Change CS_REPORTING
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:685
> (sofia/gw1/89210000000) State REPORTING
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:79
> sofia/gw1/89210000000 Standard REPORTING, cause: USER_BUSY
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:685
> (sofia/gw1/89210000000) State REPORTING going to sleep
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_state_machine.c:410
> (sofia/gw1/89210000000) State Change CS_REPORTING -> CS_DESTROY
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1229 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:13.652424 [DEBUG] switch_core_session.c:1429 Session 779
> (sofia/gw1/89210000000) Locked, Waiting on external entities
> 2012-09-13 13:46:13.672425 [DEBUG] switch_ivr_bridge.c:501
> sofia/gw1/89210000000 ending bridge by request from write function
> 2012-09-13 13:46:13.672425 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD
> DONE [sofia/onnet/78120000000 at y.y.y.y]
> 2012-09-13 13:46:13.672425 [DEBUG] switch_ivr_bridge.c:613 Send signal
> sofia/gw1/89210000000 [BREAK]
> 2012-09-13 13:46:13.672425 [DEBUG] switch_core_session.c:778 Send signal
> sofia/onnet/78120000000 at y.y.y.y [BREAK]
> EXECUTE sofia/onnet/78120000000 at y.y.y.ybridge(sofia/gateway/gw2/89210000000)
>
>
> Regards,
> Kirill
>
>
> 13.09.2012 12:55, David Villasmil ?????:
>
> Hello guys,
>
>  the way you have it:
>
>  <action application="set" data="continue_on_fail=1,2,3,
> 6,25,34,38,41,42,44,47,63,66,500,501"/>
>
>  Never worked for me, I had to set it up like:
>
>  <action application="set" data="
> continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER"/>
>
>
>  With the description text, not the error number...
>
>  Hope this helps you..
>
>
>  David
>
> On Wed, Sep 12, 2012 at 3:53 PM, iam <iam at onnet.su> wrote:
>
>> Hi Vitaly!
>>
>> Did you ever solve this problem?
>> I have the same continue_on_fail behaviour.
>>
>> http://jira.freeswitch.org/browse/FS-4232
>>
>> Vitaly added a comment - 26/Jun/12 3:14 AM
>> Sorry for the long absence, but the problem is still actual for me.
>> I replaced in my dialplan continue_on_fail to "true", but it's not give
>> desired effect.
>> May be some additional variables can help me?
>>
>> Regards,
>> Kirill
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.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.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 vitaliy.davudov at vts24.ru  Thu Sep 13 14:52:27 2012
From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=)
Date: Thu, 13 Sep 2012 14:52:27 +0400
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <1347457994946-7582782.post@n2.nabble.com>
References: <4F87EF60.3070105@vts24.ru>
	<CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
	<4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru>
	<CAKzWOxVs4qWH2ECOORXDQob5AOCrznitJabvLj+N+GGZ4Dzjig@mail.gmail.com>
	<4FAB719E.9010800@vts24.ru> <4FB4C69A.7070108@vts24.ru>
	<CANBhyPB2dZuAqG3CraDP=Hv8VU-c_MbF8AMOuruVpicQh_5YyQ@mail.gmail.com>
	<1347457994946-7582782.post@n2.nabble.com>
Message-ID: <5051BAEB.6010102@vts24.ru>

Hi, Kirill!

I solved it by adding lua-script after "bridge" application in my dialplan:

endpoint_disposition = session:getVariable("endpoint_disposition")
last_bridge_hangup_cause = session:getVariable("last_bridge_hangup_cause")
if (last_bridge_hangup_cause == "USER_BUSY" and endpoint_disposition == 
"EARLY MEDIA" ) then
     session:hangup();
end

You can see these variables by adding "info" application in the 
corresponding extension.

12.09.2012 17:53, iam ?????:
> Hi Vitaly!
>
> Did you ever solve this problem?
> I have the same continue_on_fail behaviour.
>
> http://jira.freeswitch.org/browse/FS-4232
>
> Vitaly added a comment - 26/Jun/12 3:14 AM
> Sorry for the long absence, but the problem is still actual for me.
> I replaced in my dialplan continue_on_fail to "true", but it's not give
> desired effect.
> May be some additional variables can help me?
>
> Regards,
> Kirill
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.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 iam at onnet.su  Thu Sep 13 15:13:21 2012
From: iam at onnet.su (Kirill Sysoev)
Date: Thu, 13 Sep 2012 15:13:21 +0400
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <5051BAEB.6010102@vts24.ru>
References: <4F87EF60.3070105@vts24.ru>
	<CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
	<4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru>
	<CAKzWOxVs4qWH2ECOORXDQob5AOCrznitJabvLj+N+GGZ4Dzjig@mail.gmail.com>
	<4FAB719E.9010800@vts24.ru> <4FB4C69A.7070108@vts24.ru>
	<CANBhyPB2dZuAqG3CraDP=Hv8VU-c_MbF8AMOuruVpicQh_5YyQ@mail.gmail.com>
	<1347457994946-7582782.post@n2.nabble.com>
	<5051BAEB.6010102@vts24.ru>
Message-ID: <5051BFD1.7030504@onnet.su>

Hi David, Vitaly!

Thank you for your answers.
Will try to implement your advices.

Best,
Kirill

13.09.2012 14:52, ??????? ??????? ?????:
> Hi, Kirill!
>
> I solved it by adding lua-script after "bridge" application in my 
> dialplan:
>
> endpoint_disposition = session:getVariable("endpoint_disposition")
> last_bridge_hangup_cause = 
> session:getVariable("last_bridge_hangup_cause")
> if (last_bridge_hangup_cause == "USER_BUSY" and endpoint_disposition 
> == "EARLY MEDIA" ) then
>     session:hangup();
> end
>
> You can see these variables by adding "info" application in the 
> corresponding extension.
>
> 12.09.2012 17:53, iam ?????:
>> Hi Vitaly!
>>
>> Did you ever solve this problem?
>> I have the same continue_on_fail behaviour.
>>
>> http://jira.freeswitch.org/browse/FS-4232
>>
>> Vitaly added a comment - 26/Jun/12 3:14 AM
>> Sorry for the long absence, but the problem is still actual for me.
>> I replaced in my dialplan continue_on_fail to "true", but it's not give
>> desired effect.
>> May be some additional variables can help me?
>>
>> Regards,
>> Kirill
>>
>>
>>
>> -- 
>> View this message in context: 
>> http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.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 mi.ke at null.net  Thu Sep 13 17:58:25 2012
From: mi.ke at null.net (Mi Ke)
Date: Thu, 13 Sep 2012 09:58:25 -0400
Subject: [Freeswitch-users] Variable continue on fail
Message-ID: <20120913135826.6840@gmx.com>

Hi Kirill,

 We use the following construction:

 <extension name="ex1">
 <condition field="destination_number" expression="your_dest_num_here">
 <action application="set" data="bridge_answer_timeout=120"/>
 <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,CALL_REJECTED,RECOVERY_ON_TIMER_EXPIRE"/>
 <action application="set" data="fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL"/> <---this prevents further failover in appropriate cases
 <action application="set" data="bridge_early_media=true"/>
 <action application="set" data="hangup_after_bridge=true"/>
 <action application="lua" data="get_dial_string.lua"/>
 <action application="bridge" data="${out_dial_string}"/>
 <anti-action application="pre_answer"/>
 <anti-action application="playback" data="/usr/local/freeswitch/recordings/number_incorrect.mp3"/>
 <anti-action application="hangup" data="INVALID_NUMBER_FORMAT"/>
 </condition>
 </extension>

 where get_dial_string app returns corresponding dialpeers as [dp1_params]sofia/external/dp1|[dp2_params]sofia/external/dp2|[dp3_params]sofia/external/dp3

 It works without any issues.

 FreeSWITCH Version 1.2.0-rc2+git~20120815T215213Z~c6d7417aec, CentOS 6.2 x64

 HTH,
 Mike

----- Original Message -----
From: Kirill Sysoev
Sent: 09/13/12 02:13 PM
Subject: Re: [Freeswitch-users] Variable continue on fail

 Hi David, Vitaly! Thank you for your answers. Will try to implement your advices. Best, Kirill 13.09.2012 14:52, ??????? ??????? ?????: > Hi, Kirill! > > I solved it by adding lua-script after "bridge" application in my > dialplan: > > endpoint_disposition = session:getVariable("endpoint_disposition") > last_bridge_hangup_cause = > session:getVariable("last_bridge_hangup_cause") > if (last_bridge_hangup_cause == "USER_BUSY" and endpoint_disposition > == "EARLY MEDIA" ) then > session:hangup(); > end > > You can see these variables by adding "info" application in the > corresponding extension. > > 12.09.2012 17:53, iam ?????: >> Hi Vitaly! >> >> Did you ever solve this problem? >> I have the same continue_on_fail behaviour. >> >> http://jira.freeswitch.org/browse/FS-4232 >> >> Vitaly added a comment - 26/Jun/12 3:14 AM >> Sorry for the long absence, but the problem is still actual for me. >> I replaced in my dialplan continue_on_fail to "true", but it's not give >> desired effect. >> May be some additional variables can help me? >> >> Regards, >> Kirill >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.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
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From avi at avimarcus.net  Thu Sep 13 18:19:25 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 13 Sep 2012 17:19:25 +0300
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <20120913135826.6840@gmx.com>
References: <20120913135826.6840@gmx.com>
Message-ID: <CALNZuwkKHXcmgsxySJ0OV5Qg8KNCUj+1CtcZTwUM9cS9fqVVtA@mail.gmail.com>

Can someone update the wiki
http://wiki.freeswitch.org/wiki/Variable_continue_on_fail on this?
I haven't quite figure out how to use this var.

Thanks,
-Avi


On Thu, Sep 13, 2012 at 4:58 PM, Mi Ke <mi.ke at null.net> wrote:

> Hi Kirill,
>
> We use the following construction:
>
>         <extension name="ex1">
>             <condition field="destination_number"
> expression="your_dest_num_here">
>                 <action application="set"
> data="bridge_answer_timeout=120"/>
>                 <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,CALL_REJECTED,RECOVERY_ON_TIMER_EXPIRE"/>
>                 <action application="set"
> data="fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL"/>
> <---this prevents further failover in appropriate cases
>                 <action application="set" data="bridge_early_media=true"/>
>                 <action application="set" data="hangup_after_bridge=true"/>
>                 <action application="lua" data="get_dial_string.lua"/>
>                 <action application="bridge" data="${out_dial_string}"/>
>                 <anti-action application="pre_answer"/>
>                 <anti-action application="playback"
> data="/usr/local/freeswitch/recordings/number_incorrect.mp3"/>
>                 <anti-action application="hangup"
> data="INVALID_NUMBER_FORMAT"/>
>             </condition>
>         </extension>
>
> where get_dial_string app returns corresponding dialpeers as
> [dp1_params]sofia/external/dp1|[dp2_params]sofia/external/dp2|[dp3_params]sofia/external/dp3
>
> It works without any issues.
>
> FreeSWITCH Version 1.2.0-rc2+git~20120815T215213Z~c6d7417aec, CentOS 6.2
> x64
>
> HTH,
> Mike
>
>
>
>  ----- Original Message -----
>
> From: Kirill Sysoev
>
> Sent: 09/13/12 02:13 PM
>
> Subject: Re: [Freeswitch-users] Variable continue on fail
>
>
> Hi David, Vitaly!
>
> Thank you for your answers.
> Will try to implement your advices.
>
> Best,
> Kirill
>
> 13.09.2012 14:52, ??????? ??????? ?????:
> > Hi, Kirill!
> >
> > I solved it by adding lua-script after "bridge" application in my
> > dialplan:
> >
> > endpoint_disposition = session:getVariable("endpoint_disposition")
> > last_bridge_hangup_cause =
> > session:getVariable("last_bridge_hangup_cause")
> > if (last_bridge_hangup_cause == "USER_BUSY" and endpoint_disposition
> > == "EARLY MEDIA" ) then
> >     session:hangup();
> > end
> >
> > You can see these variables by adding "info" application in the
> > corresponding extension.
> >
> > 12.09.2012 17:53, iam ?????:
> >> Hi Vitaly!
> >>
> >> Did you ever solve this problem?
> >> I have the same continue_on_fail behaviour.
> >>
> >> http://jira.freeswitch.org/browse/FS-4232
> >>
> >> Vitaly added a comment - 26/Jun/12 3:14 AM
> >> Sorry for the long absence, but the problem is still actual for me.
> >> I replaced in my dialplan continue_on_fail to "true", but it's not give
> >> desired effect.
> >> May be some additional variables can help me?
> >>
> >> Regards,
> >> Kirill
> >>
> >>
> >>
> >> --
> >> View this message in context:
> >> http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From iam at onnet.su  Thu Sep 13 18:42:42 2012
From: iam at onnet.su (Kirill Sysoev)
Date: Thu, 13 Sep 2012 18:42:42 +0400
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <CALNZuwkKHXcmgsxySJ0OV5Qg8KNCUj+1CtcZTwUM9cS9fqVVtA@mail.gmail.com>
References: <20120913135826.6840@gmx.com>
	<CALNZuwkKHXcmgsxySJ0OV5Qg8KNCUj+1CtcZTwUM9cS9fqVVtA@mail.gmail.com>
Message-ID: <5051F0E2.4020706@onnet.su>

Hi all!

Really helpful and interesting answers.
I can confirm that Vitaliy's lua script alreafy works like a charm for me.
fail_on_single_reject variable wasn't even noticed by me, while I was 
looking for a solution.
It seems like it exists exactly for such situations...

Thank you again!

Regards,

13.09.2012 18:19, Avi Marcus ?????:
> Can someone update the wiki 
> http://wiki.freeswitch.org/wiki/Variable_continue_on_fail on this?
> I haven't quite figure out how to use this var.
>
> Thanks,
> -Avi
>
>
> On Thu, Sep 13, 2012 at 4:58 PM, Mi Ke <mi.ke at null.net 
> <mailto:mi.ke at null.net>> wrote:
>
>     Hi Kirill,
>
>     We use the following construction:
>
>             <extension name="ex1">
>                 <condition field="destination_number"
>     expression="your_dest_num_here">
>                     <action application="set"
>     data="bridge_answer_timeout=120"/>
>                     <action application="set"
>     data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,CALL_REJECTED,RECOVERY_ON_TIMER_EXPIRE"/>
>                     <action application="set"
>     data="fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL"/>
>     <---this prevents further failover in appropriate cases
>                     <action application="set"
>     data="bridge_early_media=true"/>
>                     <action application="set"
>     data="hangup_after_bridge=true"/>
>                     <action application="lua" data="get_dial_string.lua"/>
>                     <action application="bridge"
>     data="${out_dial_string}"/>
>                     <anti-action application="pre_answer"/>
>                     <anti-action application="playback"
>     data="/usr/local/freeswitch/recordings/number_incorrect.mp3"/>
>                     <anti-action application="hangup"
>     data="INVALID_NUMBER_FORMAT"/>
>                 </condition>
>             </extension>
>
>     where get_dial_string app returns corresponding dialpeers as
>     [dp1_params]sofia/external/dp1|[dp2_params]sofia/external/dp2|[dp3_params]sofia/external/dp3
>
>     It works without any issues.
>
>     FreeSWITCH Version 1.2.0-rc2+git~20120815T215213Z~c6d7417aec,
>     CentOS 6.2 x64
>
>     HTH,
>     Mike
>
>>     ----- Original Message -----
>>
>>     From: Kirill Sysoev
>>
>>     Sent: 09/13/12 02:13 PM
>>
>>     Subject: Re: [Freeswitch-users] Variable continue on fail
>>
>>
>>       
>>     Hi David, Vitaly!
>>
>>     Thank you for your answers.
>>     Will try to implement your advices.
>>
>>     Best,
>>     Kirill
>>
>>     13.09.2012 14:52, ??????? ??????? ?????:
>>     > Hi, Kirill!
>>     >
>>     > I solved it by adding lua-script after "bridge" application in my
>>     > dialplan:
>>     >
>>     > endpoint_disposition = session:getVariable("endpoint_disposition")
>>     > last_bridge_hangup_cause =
>>     > session:getVariable("last_bridge_hangup_cause")
>>     > if (last_bridge_hangup_cause == "USER_BUSY" and endpoint_disposition
>>     > == "EARLY MEDIA" ) then
>>     >     session:hangup();
>>     > end
>>     >
>>     > You can see these variables by adding "info" application in the
>>     > corresponding extension.
>>     >
>>     > 12.09.2012 17:53, iam ?????:
>>     >> Hi Vitaly!
>>     >>
>>     >> Did you ever solve this problem?
>>     >> I have the same continue_on_fail behaviour.
>>     >>
>>     >>http://jira.freeswitch.org/browse/FS-4232  
>>     >>
>>     >> Vitaly added a comment - 26/Jun/12 3:14 AM
>>     >> Sorry for the long absence, but the problem is still actual for me.
>>     >> I replaced in my dialplan continue_on_fail to "true", but it's not give
>>     >> desired effect.
>>     >> May be some additional variables can help me?
>>     >>
>>     >> Regards,
>>     >> Kirill
>>     >>
>>     >>
>>     >>
>>     >> --
>>     >> View this message in context:
>>     >>http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.html  
>>     >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>     >>
>>     >> _________________________________________________________________________
>>     >>
>>     >> 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
>
>
>     _________________________________________________________________________
>     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
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Thu Sep 13 19:20:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Sep 2012 08:20:28 -0700
Subject: [Freeswitch-users] Bridge after Leg B has connected
In-Reply-To: <CAFvodxU+0agXtS-NsH5EWRJhW5yecQE_XbgvgjVu6Kqf0um6qA@mail.gmail.com>
References: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
	<CAKzWOxUgXFMTEVH_6Zx55Ga2CSQNmQuEsLPOFpGBan85MM3Z4g@mail.gmail.com>
	<CAFvodxU+0agXtS-NsH5EWRJhW5yecQE_XbgvgjVu6Kqf0um6qA@mail.gmail.com>
Message-ID: <CAKzWOxXbygTfg_0OYBZZd=kwCS_-To8J-B6+FRqmeizYu7Z2CQ@mail.gmail.com>

Perhaps this one:
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_hangup_cause

-MC

On Wed, Sep 12, 2012 at 8:36 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi,
>
> Is there any variable to get the information about the call b leg status?
>
> Because based on the call b leg hangup cause we are trying to do call
> retry mechanism in our app.
> On Sep 12, 2012 8:43 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
>>
>>
>> On Wed, Sep 12, 2012 at 2:21 AM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>>
>>> Hi,
>>>
>>>  I have a simple dialplan that just do bridge for incoming calls. I'm
>>> trying to do bridge the Call Leg A with Call Leg B after getting
>>> confirmation of call connected from Call Leg B
>>>
>>>   <action application="bridge" data="(sofia or
>>> freetdm)/internal/destination number"/> **
>>>
>>> Is there any way to bridge the both the call legs after getting
>>> confirmation from call leg B?
>>>
>> The legs will be bridged as soon as the B leg returns media (SIP 180/183)
>> or in the case of ignoring early media it bridges when receiving a SIP 200
>> OK. In other words, the bridge app does all the work for you.
>>
>> Now, if you want the called party to press a key to acknowledge receipt
>> of the call then you need the 'answer confirmation' feature:
>>
>> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>>
>> HtH,
>> MC
>>
>>>
>>> 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
>>>
>>>
>>
>>
>> --
>> 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 gvvsubhashkumar at gmail.com  Thu Sep 13 19:36:32 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Thu, 13 Sep 2012 08:36:32 -0700
Subject: [Freeswitch-users] Bridge after Leg B has connected
In-Reply-To: <CAKzWOxXbygTfg_0OYBZZd=kwCS_-To8J-B6+FRqmeizYu7Z2CQ@mail.gmail.com>
References: <CAFvodxU7cypmcOWoLe-pTGNmt+psZeAGg9fXwwqCTVj+LFrEoQ@mail.gmail.com>
	<CAKzWOxUgXFMTEVH_6Zx55Ga2CSQNmQuEsLPOFpGBan85MM3Z4g@mail.gmail.com>
	<CAFvodxU+0agXtS-NsH5EWRJhW5yecQE_XbgvgjVu6Kqf0um6qA@mail.gmail.com>
	<CAKzWOxXbygTfg_0OYBZZd=kwCS_-To8J-B6+FRqmeizYu7Z2CQ@mail.gmail.com>
Message-ID: <CAFvodxXciPiGzVqFqDtksZYDVkrETR36L5BbA=jHaAcqDOAe=A@mail.gmail.com>

Thanks,

Can we use it in our dialplan? If yes how can we check it before bridging a
call.

Thanks in advanvce...

Thanks,
Subhash.
On Sep 13, 2012 8:55 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Perhaps this one:
> http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_hangup_cause
>
> -MC
>
> On Wed, Sep 12, 2012 at 8:36 AM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>
>> Hi,
>>
>> Is there any variable to get the information about the call b leg status?
>>
>> Because based on the call b leg hangup cause we are trying to do call
>> retry mechanism in our app.
>> On Sep 12, 2012 8:43 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>>
>>>
>>>
>>> On Wed, Sep 12, 2012 at 2:21 AM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>>>
>>>> Hi,
>>>>
>>>>  I have a simple dialplan that just do bridge for incoming calls. I'm
>>>> trying to do bridge the Call Leg A with Call Leg B after getting
>>>> confirmation of call connected from Call Leg B
>>>>
>>>>   <action application="bridge" data="(sofia or
>>>> freetdm)/internal/destination number"/> **
>>>>
>>>> Is there any way to bridge the both the call legs after getting
>>>> confirmation from call leg B?
>>>>
>>> The legs will be bridged as soon as the B leg returns media (SIP
>>> 180/183) or in the case of ignoring early media it bridges when receiving a
>>> SIP 200 OK. In other words, the bridge app does all the work for you.
>>>
>>> Now, if you want the called party to press a key to acknowledge receipt
>>> of the call then you need the 'answer confirmation' feature:
>>>
>>> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>>>
>>> HtH,
>>> MC
>>>
>>>>
>>>> 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
>>>>
>>>>
>>>
>>>
>>> --
>>> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
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From lists at kavun.ch  Thu Sep 13 21:30:06 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Sep 2012 13:30:06 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <27905.1347525413@ccs.covici.com>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
Message-ID: <A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>

To clarify further:

The DTMFs are capture when the invalid prompt is played and only executed after the prompt ends, as if they were queued.

To reproduce:

Execute the play_and_get_digits app with an invalid prompt of something a little longer than 5 secondes, just to give you enough time to enter some digits.
When the initial prompt plays, dial some digits that do not match your regex so that you end up in the invalid prompt.
While the invalid prompt is playing, try dialing some DTMFs as if you had made a mistake and were dialing the correct sequence this time around.
Even if you key "#", the invalid prompt will still be played till the end and all your digits queued for execution right after.

Thanks for testing,
Emrah
On Sep 13, 2012, at 4:36 AM, covici at ccs.covici.com wrote:

> I am not seeing this -- I can hit the dtmf while the prompt is still
> playing.  When its playing the invalid prompt, what could you type
> anyway -- it may not be listening yet.
> 
> Emrah <lists at kavun.ch> wrote:
> 
>> This happens with the stable version as well. It seems to be limited to the play_and_get_digits app though.
>> 
>> Best,
>> Emrah
>> On Sep 13, 2012, at 2:34 AM, Emrah <lists at kavun.ch> wrote:
>> 
>>> Thanks Michael. I will check out the stable version and let you know.
>>> 
>>> Regards,
>>> Emrah
>>> On Sep 11, 2012, at 8:02 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> 
>>>> I do not see this behavior on 1.2.stable branch. When I press a key during the invalid prompt it breaks out of playback immediately and attempts to process whatever digit(s) I input. I tested with x5000 sample IVR:
>>>> Dial 5000
>>>> Press 7, wait for "invalid entry" message to start playing
>>>> Press 3 while she's saying "that was an invalid entry"
>>>> After a few seconds MOH comes on.
>>>> 
>>>> -MC
>>>> 
>>>> On Tue, Sep 11, 2012 at 12:43 PM, Emrah <lists at kavun.ch> wrote:
>>>> Just wanted to drop a quick note about an issue I just stumbled upon.
>>>> 
>>>> When using play_and_get_digits, you are still able to input DTMFs when the invalid-entry prompt is played, but the message won't be interrupted and your DTMF input won't be executed until the prompt is fully played.
>>>> 
>>>> Can someone look into this and confirm?
>>>> 
>>>> 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
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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



From lists at kavun.ch  Thu Sep 13 23:09:25 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Sep 2012 15:09:25 -0400
Subject: [Freeswitch-users] 481 Call Leg/Transaction Does Not Exist
Message-ID: <5D5FD0B1-34C0-4C06-AD20-735E3ACE8803@kavun.ch>

Hi all,

I seem to be encountering the strangest problems.
I have been doing some more testing with my transfers and here is a recurring issue.

I have 2 phones behind the same router. 1 Polycom VVX 1500 and 1 Yealink SIP-T38g
They are both registering on the same FS server which is on a remote public IP.
The Polycom has extension 20 and the yealink uses 21.
When I call the Polycom from the Yealink and perform a blind transfer from the Yealink, it fails with: 481 Call Leg/Transaction Does Not Exist.
If I transfer from the Polycom though, it seems to work.

In resume:
I cannot transfer from A leg. B leg doesn't seem to know its way around and looses it. To demonstrate, I transfer B leg from A leg to extension 5000.
http://pastebin.freeswitch.org/19893

I tried Googleing the error code and didn't find anything particularly related to my issue.

Any hint would be greatly appreciated. To make sure, I downgraded to the stable version.

Thanks a bunch, as always.
Emrah

From TamurlangClan at gmail.com  Thu Sep 13 23:57:47 2012
From: TamurlangClan at gmail.com (TamurlangClan)
Date: Thu, 13 Sep 2012 12:57:47 -0700 (PDT)
Subject: [Freeswitch-users]  Kazoo Review
Message-ID: <34429093.post@talk.nabble.com>


Hi,

I'm thinking to use Kazoo by 2600HZ for effectively mainting PBX and other
services. Has anyone used it? Anyone willing to give feedback on how
effective it is? Thanks.
-- 
View this message in context: http://old.nabble.com/Kazoo-Review-tp34429093p34429093.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From tamurlangclan at gmail.com  Thu Sep 13 23:58:35 2012
From: tamurlangclan at gmail.com (Tamurlang Clan)
Date: Fri, 14 Sep 2012 01:28:35 +0530
Subject: [Freeswitch-users] Easy Information Needed For Placement Of POP
 To Avoid Latency
In-Reply-To: <CAO47HpcLOz3hL6gv=6PcM+YnfzJXss-eSm22vYh5KBMmaB6k8Q@mail.gmail.com>
References: <34410404.post@talk.nabble.com>
	<CAO47HpcLOz3hL6gv=6PcM+YnfzJXss-eSm22vYh5KBMmaB6k8Q@mail.gmail.com>
Message-ID: <CAHbF-TVfC23ahYA-ZAP2o487hQ-z1Kb1vELvTpK6qO+RocR9vg@mail.gmail.com>

Thanks alot.

Yeah sure, I will explore your services :).

On Mon, Sep 10, 2012 at 5:28 PM, jay binks <jaybinks at gmail.com> wrote:

> you are 100% correct... you do not want your traffic routed through the US
> if your users are in Australia.
>
> also you are always better to choose a provider with the shortest path to
> you ( most direct )
> because of latency, packet loss and jitter.
>
> <shameless plug>
> if your users truly are in Australia you should select the BEST SIP
> trunking provider in Australia.
> http://netsip.com.au/
>
> We are also supporters and users of Freeswitch, so you also help the
> community by signing up with us :)
> </shameless plug>
>
>
>
>
>
> On 10 September 2012 16:09, TamurlangClan <TamurlangClan at gmail.com> wrote:
>
>>
>> Hi,
>>
>> I am weak in network architecture and need help in placement of POP.
>> Please
>> see below:
>>
>> 1) Is it must to have a POP close to where your subsicriber base. For
>> Example: If my user base is in Australia as well, Will they be affected if
>> their calls are routed through POP in New York as it might cause Latency
>> and
>> packet delay thereby dropping call quality considerably?
>>
>> 2) I have seen some termination providers who have indirect route. Does
>> this
>> mean if i choose them my calls will be routed through multiple switches at
>> multiple locations which will give poor quality voice and also high
>> Latency
>> and Packet Delay? So is it better choose direct routes?
>>
>> Please guide on this. Thanks in advance.
>> --
>> View this message in context:
>> http://old.nabble.com/Easy-Information-Needed-For-Placement-Of-POP-To-Avoid-Latency-tp34410404p34410404.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
>>
>
>
>
> --
> 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
>
>
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From msc at freeswitch.org  Fri Sep 14 01:36:33 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Sep 2012 14:36:33 -0700
Subject: [Freeswitch-users] Kazoo Review
In-Reply-To: <34429093.post@talk.nabble.com>
References: <34429093.post@talk.nabble.com>
Message-ID: <CAKzWOxXfHkTjQF+9sQUG2sXhCtpO_H=88GWjp1dGb3KEo+TjcQ@mail.gmail.com>

You are better off talking to the 2600hz community on this one. There are a
scant handful of people here who have used Kazoo but if you get on the
2600hz group list you'll get to the target demographic you are seeking.

-MC

On Thu, Sep 13, 2012 at 12:57 PM, TamurlangClan <TamurlangClan at gmail.com>wrote:

>
> Hi,
>
> I'm thinking to use Kazoo by 2600HZ for effectively mainting PBX and other
> services. Has anyone used it? Anyone willing to give feedback on how
> effective it is? Thanks.
> --
> View this message in context:
> http://old.nabble.com/Kazoo-Review-tp34429093p34429093.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jpablolorenzetti at hotmail.com  Fri Sep 14 02:10:48 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 13 Sep 2012 22:10:48 +0000
Subject: [Freeswitch-users] calls remain connected after conference
	initiator hangs up
Message-ID: <DUB002-W1529765E14C38AEFFB295A8D9910@phx.gbl>

Hi, i have a little problem that i m sure it is easy to fix , my extensions make conferences (the confonference button on the phone) and/or transfers and then they hag up but the others in the conference remainconnected, so this is provoking some people to abuse by connecting their friends through the company PBX and then they hang up and leave the friendsconnected in the call ... i have not succeeded so far in finding how to fix this. 
Also, it is possible to allow to transfer calls from outside to inside and/or inside to outsidebut not to transfer calls from outside to outside ?thanks! 		 	   		  
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From mike.burlingame at me.com  Fri Sep 14 02:49:57 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Thu, 13 Sep 2012 15:49:57 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
Message-ID: <64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>

So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled  
basic flow is
 
                        FS Server A                                                      FS Server B
|-----------------DID Channel---------------------|     |---------Termination Channel (RCF)----------|
Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier

Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)

FS Server A Logs
http://pastebin.freeswitch.org/19897

FS Server B SIP Logs
http://pastebin.freeswitch.org/19896






On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:

> 
> 
> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
> Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
> 
> Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared 
> 
> Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
> 
> -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 Sep 14 02:58:38 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Sep 2012 17:58:38 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <64F1449C-A35E-48E0-88F6-595C18F7D0DC@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
Message-ID: <CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>

sigh, you forgot "console loglevel debug"  this is only info
loglevel.........


On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> So it seems all is not well - missing ACK and forward of 200OK when 2 FS
> servers are in the mix when sip_wait_for_aleg_ack=true is enabled
> basic flow is
>
>                         FS Server A
>                FS Server B
> |-----------------DID Channel---------------------|
> |---------Termination Channel (RCF)----------|
> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs
> --> Carrier
>
> Both FS running latest GIT as of this morning - Both FS servers
> using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not
> getting an ACK back from FS_A)
>
> FS Server A Logs
> http://pastebin.freeswitch.org/19897
>
> FS Server B SIP Logs
> http://pastebin.freeswitch.org/19896
>
>
>
>
>
>
> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
>
> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> Crash was resolved this weekend by a code update via jira case - no more
>> crashes after the update - some more dial plan edits and a little more
>> testing I will be ready to head over to pay the Wiki tax
>>
>> Thanks guys for adding this feature it seems to be working as expected
>> and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
>>
>
> Thanks for checking in and letting us know where you stand. I really like
> it when these threads have some sort of resolution. If you have any issues
> with wiki editing please let me know or hop on IRC for an assist.
>
> -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
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From mike.burlingame at me.com  Fri Sep 14 03:02:14 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Thu, 13 Sep 2012 16:02:14 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
Message-ID: <3C8A32C6-874F-4768-9AA9-EDCB98F42EF3@me.com>

I currently have 2 servers out of production right now I can have Opensips redirect calls to the two servers for my ANI and the Dest number and re-run the output 


On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> sigh, you forgot "console loglevel debug"  this is only info loglevel.........
> 
> 
> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled  
> basic flow is
>  
>                         FS Server A                                                      FS Server B
> |-----------------DID Channel---------------------|     |---------Termination Channel (RCF)----------|
> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
> 
> Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
> 
> FS Server A Logs
> http://pastebin.freeswitch.org/19897
> 
> FS Server B SIP Logs
> http://pastebin.freeswitch.org/19896
> 
> 
> 
> 
> 
> 
> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> 
>> 
>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
>> 
>> Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared 
>> 
>> Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
>> 
>> -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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From djbinter at gmail.com  Fri Sep 14 03:07:31 2012
From: djbinter at gmail.com (DJB International)
Date: Thu, 13 Sep 2012 16:07:31 -0700
Subject: [Freeswitch-users] T.38 with Re-INVITE question
Message-ID: <CAAB3jWuXq67RX6j3=Zcv=9H432Ot=JkrrfMfrUVdi99Gf-VSPA@mail.gmail.com>

I was testing T.38 with Re-INVITE.

Call Flow:  Inbound Carrier (1.2.3.4) -> FS (Bypass Media) -> Outbound
Carrier that will send T.38 Re-INVITE (6.7.8.9).

However, I noticed that FS generated 2 CDRs for the same call, which are:
1) A-leg CDR from inbound carrier, and also 2) A-leg CDR from outbound
carrier that send T.38 Re-INVITE.

I am wondering whether this is a normal behavior, or there is a way to
ignore the 2nd CDR?


Example of 2 FS CDRs below:


1 : caller_id_name                      = +16193301897
2 : caller_id_number                    = +16193301897
3 : aniii                               = 0
4 : destination_number                  = +12133456789
5 : context                             = routing
6 : start_stamp                         = 2012-09-13 21:12:13
7 : answer_stamp                        = 2012-09-13 21:12:13
8 : end_stamp                           = 2012-09-13 21:14:06
9 : duration                            = 113
10: billsec                             = 113
11: hangup_cause                        = NORMAL_CLEARING
12: channel_name                        = sofia/fs/+16193301897 at 1.2.3.4:5060
13: bridge_channel                      =
14: network_addr                        = 1.2.3.4
15: remote_ip_last_arg                  = sofia/fs/2133456789 at 6.7.8.9
16: bleg_network_addr                   = 6.7.8.9
17: sip_hangup_disposition              = recv_bye
18: endpoint_disposition                = ANSWER

*****************************************

1 : caller_id_name                      = 2133456789
2 : caller_id_number                    = 2133456789
3 : aniii                               =
4 : destination_number                  = mod_sofia
5 : context                             = routing
6 : start_stamp                         = 2012-09-13 21:12:13
7 : answer_stamp                        = 2012-09-13 21:12:13
8 : end_stamp                           = 2012-09-13 21:14:06
9 : duration                            = 113
10: billsec                             = 113
11: hangup_cause                        = NORMAL_CLEARING
12: channel_name                        = sofia/fs/2133456789 at 6.7.8.9
13: bridge_channel                      =
14: network_addr                        = 6.7.8.9
15: remote_ip_last_arg                  =
16: bleg_network_addr                   =
17: sip_hangup_disposition              = send_bye
18: endpoint_disposition                = INBOUND CALL


Thank you.
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From mike.burlingame at me.com  Fri Sep 14 03:49:01 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Thu, 13 Sep 2012 16:49:01 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
Message-ID: <088228C8-3C8F-455A-9280-4D7B371DC1C2@me.com>

So what I am seeing is if the call flow is like

Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine

but if the flow is 

Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected 

FS_SERVER_A 
http://pastebin.freeswitch.org/19898

FS_SERVER_B
http://pastebin.freeswitch.org/19899

On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> sigh, you forgot "console loglevel debug"  this is only info loglevel.........
> 
> 
> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled  
> basic flow is
>  
>                         FS Server A                                                      FS Server B
> |-----------------DID Channel---------------------|     |---------Termination Channel (RCF)----------|
> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
> 
> Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
> 
> FS Server A Logs
> http://pastebin.freeswitch.org/19897
> 
> FS Server B SIP Logs
> http://pastebin.freeswitch.org/19896
> 
> 
> 
> 
> 
> 
> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> 
>> 
>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
>> 
>> Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared 
>> 
>> Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
>> 
>> -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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tamurlangclan at gmail.com  Fri Sep 14 10:29:31 2012
From: tamurlangclan at gmail.com (Tamurlang Clan)
Date: Fri, 14 Sep 2012 11:59:31 +0530
Subject: [Freeswitch-users] Kazoo Review
In-Reply-To: <CAKzWOxXfHkTjQF+9sQUG2sXhCtpO_H=88GWjp1dGb3KEo+TjcQ@mail.gmail.com>
References: <34429093.post@talk.nabble.com>
	<CAKzWOxXfHkTjQF+9sQUG2sXhCtpO_H=88GWjp1dGb3KEo+TjcQ@mail.gmail.com>
Message-ID: <CAHbF-TVQoDyw_gYqJcLB8vefdicX5_P+rBN2z4OCGCMhaU2jPA@mail.gmail.com>

Sorry but thanks Micheal.

On Fri, Sep 14, 2012 at 3:06 AM, Michael Collins <msc at freeswitch.org> wrote:

> You are better off talking to the 2600hz community on this one. There are
> a scant handful of people here who have used Kazoo but if you get on the
> 2600hz group list you'll get to the target demographic you are seeking.
>
> -MC
>
>
> On Thu, Sep 13, 2012 at 12:57 PM, TamurlangClan <TamurlangClan at gmail.com>wrote:
>
>>
>> Hi,
>>
>> I'm thinking to use Kazoo by 2600HZ for effectively mainting PBX and other
>> services. Has anyone used it? Anyone willing to give feedback on how
>> effective it is? Thanks.
>> --
>> View this message in context:
>> http://old.nabble.com/Kazoo-Review-tp34429093p34429093.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
>>
>
>
>
> --
> 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 gabe at gundy.org  Fri Sep 14 13:35:57 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 14 Sep 2012 03:35:57 -0600
Subject: [Freeswitch-users] T.38 with Re-INVITE question
In-Reply-To: <CAAB3jWuXq67RX6j3=Zcv=9H432Ot=JkrrfMfrUVdi99Gf-VSPA@mail.gmail.com>
References: <CAAB3jWuXq67RX6j3=Zcv=9H432Ot=JkrrfMfrUVdi99Gf-VSPA@mail.gmail.com>
Message-ID: <CAMwZ-t+f7dnfj3DakFxnfZTGi+J44Gawn3ycPi59Aktdi5uyAA@mail.gmail.com>

On Thu, Sep 13, 2012 at 5:07 PM, DJB International <djbinter at gmail.com> wrote:
> I was testing T.38 with Re-INVITE.
>
> Call Flow:  Inbound Carrier (1.2.3.4) -> FS (Bypass Media) -> Outbound
> Carrier that will send T.38 Re-INVITE (6.7.8.9).
>
> However, I noticed that FS generated 2 CDRs for the same call, which are: 1)
> A-leg CDR from inbound carrier, and also 2) A-leg CDR from outbound carrier
> that send T.38 Re-INVITE.
>
> I am wondering whether this is a normal behavior, or there is a way to
> ignore the 2nd CDR?


Does this help?

process_cdr <-- indicates how to process CDR records.

http://wiki.freeswitch.org/wiki/Channel_Variables#process_cdr


Good luck.

Gabe


From Alexander.Haugg at c4b.de  Fri Sep 14 11:24:17 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Fri, 14 Sep 2012 07:24:17 +0000
Subject: [Freeswitch-users] Consultation Call via event_socket interface
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>

Hi All,

I'm new on the mailing list.
I have a problem with a call scenario.

-          Channel A and channel B are bridged (A is my own channel and B is my calling partner)

-          Now i set channel B on hold with the command "uuid_hold xxx" and create a new channel to C with the command:

bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip',origination_caller_id_number=num at ip}sofia/external/num at ip  &park

this works correctly, the partner C answer and the channel is established.

-          Now the Problem:

I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false.

Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment.
Is there a general problem in my plan to do that?
Is it a better plan to do this over the dialplan?
The next step in this scenario is to toggle the connection A -> B and A -> C.

Thanks for your help!
Nice regards,
Alex
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From shishko69 at gmail.com  Fri Sep 14 12:33:31 2012
From: shishko69 at gmail.com (Shishko)
Date: Fri, 14 Sep 2012 10:33:31 +0200
Subject: [Freeswitch-users] Problem with SIP INVITE
Message-ID: <5052EBDB.3020904@gmail.com>

Hi,

one of my ITSP is using Cisco 2811 (IOS 12.4) as H.323 to SIP gateway 
for SIP trunk to my Freeswitch.
I noticed that it sends INVITE message without SDP, and then Freeswitch 
terminates the call.
I tried using 3pcc parameter as "proxy" and as "true", but no avail.

Is this expected behavior, and could I fix it somehow?

Here's SIP trace:

------------------------------------------------------------------------
recv 784 bytes from udp/[192.168.117.8]:51358 at 08:06:51.013018:
------------------------------------------------------------------------
    INVITE sip:100965461 at 192.168.117.9:5060 SIP/2.0
    Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
    From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    To: <sip:100965461 at 192.168.117.9>
    Date: Fri, 14 Sep 2012 08:07:14 GMT
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    Supported: 100rel,timer,replaces
    Min-SE:  1800
    Cisco-Guid: 2649593691-218042625-2288844833-673556762
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Remote-Party-ID: 
<sip:xxxxxxxxxxxx at 192.168.117.8>;party=calling;screen=yes;privacy=off
    Timestamp: 1347610034
    Contact: <sip:xxxxxxxxxxxx at 192.168.117.8:5060>
    Expires: 180
    Allow-Events: telephone-event

------------------------------------------------------------------------
send 372 bytes to udp/[192.168.117.8]:5060 at 08:06:51.013018:
------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
    From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    To: <sip:100965461 at 192.168.117.9>
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    CSeq: 101 INVITE
    Timestamp: 1347610034 0.000000
    User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
    Content-Length: 0

------------------------------------------------------------------------
send 889 bytes to udp/[192.168.117.8]:5060 at 08:06:51.036020:
------------------------------------------------------------------------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
    From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    CSeq: 101 INVITE
    Contact: <sip:mod_sofia at 192.168.117.9:5060>
    User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, conference, refer
    Min-SE: 1800
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 205

    v=0
    o=FreeSWITCH 1347586709 1347586710 IN IP4 192.168.117.9
    s=FreeSWITCH
    c=IN IP4 192.168.117.9
    t=0 0
    m=audio 23302 RTP/AVP 8 0 101 13
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
------------------------------------------------------------------------
recv 364 bytes from udp/[192.168.117.8]:51358 at 08:06:51.041020:
------------------------------------------------------------------------
    ACK sip:mod_sofia at 192.168.117.9:5060 SIP/2.0
    Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK731155
    From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
    Date: Fri, 14 Sep 2012 08:07:14 GMT
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    Max-Forwards: 70
    CSeq: 101 ACK
    Content-Length: 0

------------------------------------------------------------------------
send 645 bytes to udp/[192.168.117.8]:5060 at 08:06:51.041020:
------------------------------------------------------------------------
    BYE sip:xxxxxxxxxxxx at 192.168.117.8:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.117.9;rport;branch=z9hG4bKvBN2ZB9DZKZ6c
    Max-Forwards: 70
    From: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
    To: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    CSeq: 33468941 BYE
    Contact: <sip:mod_sofia at 192.168.117.9:5060>
    User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY
    Supported: timer, precondition, path, replaces
    Reason: SIP;cause=488;text="No answer to offer"
    Content-Length: 0

------------------------------------------------------------------------
recv 426 bytes from udp/[192.168.117.8]:51358 at 08:06:51.042020:
------------------------------------------------------------------------
    BYE sip:mod_sofia at 192.168.117.9:5060 SIP/2.0
    Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
    From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
    Date: Fri, 14 Sep 2012 08:07:14 GMT
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Max-Forwards: 70
    Timestamp: 1347610034
    CSeq: 102 BYE
    Content-Length: 0

------------------------------------------------------------------------
send 387 bytes to udp/[192.168.117.8]:5060 at 08:06:51.042020:
------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
    From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    CSeq: 102 BYE
    Timestamp: 1347610034 0.000000
    User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
    Content-Length: 0

------------------------------------------------------------------------
send 510 bytes to udp/[192.168.117.8]:5060 at 08:06:51.043021:
------------------------------------------------------------------------
    SIP/2.0 480 Temporarily Unavailable
    Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
    From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    CSeq: 102 BYE
    User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY
    Supported: timer, precondition, path, replaces
    Content-Length: 0

------------------------------------------------------------------------
recv 363 bytes from udp/[192.168.117.8]:5060 at 08:06:51.046021:
------------------------------------------------------------------------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.117.9;rport;branch=z9hG4bKvBN2ZB9DZKZ6c
    From: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
    To: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
    Date: Fri, 14 Sep 2012 08:07:14 GMT
    Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    CSeq: 33468941 BYE

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

Thanks,

Denis


From peetzer at gmail.com  Fri Sep 14 13:03:58 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Fri, 14 Sep 2012 11:03:58 +0200
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
Message-ID: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>

Hi all,

Just wondering if I can add/ delete users to the db from the CLI (I'm using
ESL java library)?

See http://wiki.freeswitch.org/wiki/Function_db

>From the Dialplan this is possible, I want however to dynamically add new
users.

<action application="db" data="insert/realm/key/value"/>


In the api (mod commands) nothing is mentioned about "db"

I don't know if something like this could work with my example extension
1999, default group named default...

response = client.sendSyncApiCommand( "db", "insert/default/1999/1999");

Another option is to use mod_xml_curl and feed the dialplan or user
directory to FreeSwitch back again.

Kind regards,

Peter
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From Rob.Moore at Aeriandi.com  Fri Sep 14 15:46:56 2012
From: Rob.Moore at Aeriandi.com (Rob Moore)
Date: Fri, 14 Sep 2012 11:46:56 +0000
Subject: [Freeswitch-users] Altering From Header in SIP Invite
In-Reply-To: <49C5FCA19A8A114493EBAACA42FE5899105C87C2@1AERDCEXCHMBX1.AER.AERCO.local>
References: <49C5FCA19A8A114493EBAACA42FE5899105C87C2@1AERDCEXCHMBX1.AER.AERCO.local>
Message-ID: <49C5FCA19A8A114493EBAACA42FE5899105CA323@1AERDCEXCHMBX1.AER.AERCO.local>

Hi Everyone,

For those who are interested I found a way around this issue. I'm not sure if this is the correct way to produce this result but it worked.

The problem I had was that I could set the Freephone number I wished to  present but the P asserted ID would overwrite this with the other standard number I was attempting to send.

So to resolve the issue I had to disable all CID, use effective_caller_id_number to present my Freephone number (this adds the number to your from header) then I used sip_h_ to add the P-Asserted-Identity manually.

                                <action application="set" data="effective_caller_id_number=+448000000262"/>
                                <action application="export" data="sip_cid_type=none"/>
                                <action application="export" data="sip_h_P-Asserted-Identity= <sip:+445600000262 at fixsipbizzle.org> "/>

Hope this helps anyone else who ever has to provide this unusual setup.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Moore
Sent: 11 September 2012 19:11
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Altering From Header in SIP Invite

Hi All,

I'm having a little trouble with 'presentation numbers' with a new provider I'm in IOT with this week.

I'm trying to recreate the following Invite as the calls pass through our dialplan to this provider but there are issues with trying to get a different CLI into the From and P-Asserted-Identity headers.

When presenting a Freephone number (for example) we need to still present the regular number that would be used by this extension in the P-Asserted-Identity whilst passing the number we wish to have presented in the From header.

Currently we are not using Gateways so we cannot resort to using  <param name="caller-id-in-from" value="true"/> (although I expect this won't do what we need in this case) so I've looked at altering the channel variables  sip_from_user,sip_full_from and sip_full_uri using set and export dial plan apps but none of these seem to have any effect so I guess these variables must be read only.

I'm sure this must be simple, but can't for the life of me work out what I need to do.

Below is an extract from an example header from the provider I am trying to recreate, I've also added a copy of the Dialplan extension I am using to test.
If someone can tell me what I'm getting wrong I would really appreciate it.

Thanks

Rob


INVITE sip:+445600005262 at primarysip.barfoo.com;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: +445600000262 <sip:+445600005262 at primarysip.barfoo.com;user=phone>
From: <sip:+448000655262 at uk.foo.bar.2.net;user=phone>;tag=12544
P-Asserted-Identity: <sip:+445600655262 at uk.sdin.bt.net>
Call-ID: 1347372978-13100 at mgc-uk-998.n2<mailto:1347372978-13100 at mgc-uk-998.n2>
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE


DIAL Plan: (attempting +445600655262 in P Asserted-Id and +448000655262 in from) have commented out some things that I have tried.)
If you are worried about [sip_h_X-Gateway=4003:4] this is added to help our SBC forward calls to our different providers.

   <extension name="TC 25">
     <condition field="destination_number" expression="^10125$">
                                <action application="set" data="effective_caller_id_name=_undef_"/>
                                <action application="set" data="effective_caller_id_number=+445600005262 "/>
                                <action application="set" data="dtmf_type=rfc2833"/>
                                <action application="set" data="sip_cid_type=pid"/>
                                <!--<action application="export" data=" sip_from_user =+448000655262"/> -->
                                <!-- <action application="export" data="sip_h_from=<sip:+448000655262 at primarysip.barfoo.com>;tag=${sip_from_tag} "/> -->
                                <!--  <action application="export" data="sip_h_P-Asserted-Identity= +445600005262 <sip:+445600655262 at 172.17.7.4> "/> -->
                                <action application="info"/>
                  <action application="bridge" data="[sip_h_X-Gateway=4003:4]sofia/internal/+442920001199 at pstn.tel<mailto:sofia/internal/+442920001199 at pstn.tel>" />
     </condition>
   </extension>


Rob Moore
Telephony Systems Infrastructure Manager
Aeriandi
Aeriandi Ltd, Prama House, Banbury Road, Oxford, OX27HT
E: rob.moore at aeriandi.com<mailto:rob.moore at aeriandi.com>
W: www.aeriandi.com<http://www.aeriandi.com/>
M: +44 (0)7766 838040
T: +44 (0) 845 108 0308

[Description: Description: Description: Description: Description: Description: Description: Description: cid:image002.png at 01CC9E0C.20153A40]

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From kris at kriskinc.com  Fri Sep 14 17:32:53 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 14 Sep 2012 09:32:53 -0400
Subject: [Freeswitch-users] Problem with SIP INVITE
In-Reply-To: <5052EBDB.3020904@gmail.com>
References: <5052EBDB.3020904@gmail.com>
Message-ID: <CAKDfjgeqED7CJ_y7ATxvKzL4S=PWAKk5zmD=8cdSOeVO+4ds7A@mail.gmail.com>

With late-negotiation the offer/answer model still applies.  They
should be sending you an SDP in the ACK to your 200 OK.  They aren't.
Their stuff is broken.

On Fri, Sep 14, 2012 at 4:33 AM, Shishko <shishko69 at gmail.com> wrote:
> Hi,
>
> one of my ITSP is using Cisco 2811 (IOS 12.4) as H.323 to SIP gateway
> for SIP trunk to my Freeswitch.
> I noticed that it sends INVITE message without SDP, and then Freeswitch
> terminates the call.
> I tried using 3pcc parameter as "proxy" and as "true", but no avail.
>
> Is this expected behavior, and could I fix it somehow?
>
> Here's SIP trace:
>
> ------------------------------------------------------------------------
> recv 784 bytes from udp/[192.168.117.8]:51358 at 08:06:51.013018:
> ------------------------------------------------------------------------
>     INVITE sip:100965461 at 192.168.117.9:5060 SIP/2.0
>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     To: <sip:100965461 at 192.168.117.9>
>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     Supported: 100rel,timer,replaces
>     Min-SE:  1800
>     Cisco-Guid: 2649593691-218042625-2288844833-673556762
>     User-Agent: Cisco-SIPGateway/IOS-12.x
>     Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>     CSeq: 101 INVITE
>     Max-Forwards: 70
>     Remote-Party-ID:
> <sip:xxxxxxxxxxxx at 192.168.117.8>;party=calling;screen=yes;privacy=off
>     Timestamp: 1347610034
>     Contact: <sip:xxxxxxxxxxxx at 192.168.117.8:5060>
>     Expires: 180
>     Allow-Events: telephone-event
>
> ------------------------------------------------------------------------
> send 372 bytes to udp/[192.168.117.8]:5060 at 08:06:51.013018:
> ------------------------------------------------------------------------
>     SIP/2.0 100 Trying
>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     To: <sip:100965461 at 192.168.117.9>
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     CSeq: 101 INVITE
>     Timestamp: 1347610034 0.000000
>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>     Content-Length: 0
>
> ------------------------------------------------------------------------
> send 889 bytes to udp/[192.168.117.8]:5060 at 08:06:51.036020:
> ------------------------------------------------------------------------
>     SIP/2.0 200 OK
>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     CSeq: 101 INVITE
>     Contact: <sip:mod_sofia at 192.168.117.9:5060>
>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>     Accept: application/sdp
>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>     Supported: timer, precondition, path, replaces
>     Allow-Events: talk, hold, conference, refer
>     Min-SE: 1800
>     Content-Type: application/sdp
>     Content-Disposition: session
>     Content-Length: 205
>
>     v=0
>     o=FreeSWITCH 1347586709 1347586710 IN IP4 192.168.117.9
>     s=FreeSWITCH
>     c=IN IP4 192.168.117.9
>     t=0 0
>     m=audio 23302 RTP/AVP 8 0 101 13
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16
>     a=ptime:20
> ------------------------------------------------------------------------
> recv 364 bytes from udp/[192.168.117.8]:51358 at 08:06:51.041020:
> ------------------------------------------------------------------------
>     ACK sip:mod_sofia at 192.168.117.9:5060 SIP/2.0
>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK731155
>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     Max-Forwards: 70
>     CSeq: 101 ACK
>     Content-Length: 0
>
> ------------------------------------------------------------------------
> send 645 bytes to udp/[192.168.117.8]:5060 at 08:06:51.041020:
> ------------------------------------------------------------------------
>     BYE sip:xxxxxxxxxxxx at 192.168.117.8:5060 SIP/2.0
>     Via: SIP/2.0/UDP 192.168.117.9;rport;branch=z9hG4bKvBN2ZB9DZKZ6c
>     Max-Forwards: 70
>     From: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>     To: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     CSeq: 33468941 BYE
>     Contact: <sip:mod_sofia at 192.168.117.9:5060>
>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>     Supported: timer, precondition, path, replaces
>     Reason: SIP;cause=488;text="No answer to offer"
>     Content-Length: 0
>
> ------------------------------------------------------------------------
> recv 426 bytes from udp/[192.168.117.8]:51358 at 08:06:51.042020:
> ------------------------------------------------------------------------
>     BYE sip:mod_sofia at 192.168.117.9:5060 SIP/2.0
>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     User-Agent: Cisco-SIPGateway/IOS-12.x
>     Max-Forwards: 70
>     Timestamp: 1347610034
>     CSeq: 102 BYE
>     Content-Length: 0
>
> ------------------------------------------------------------------------
> send 387 bytes to udp/[192.168.117.8]:5060 at 08:06:51.042020:
> ------------------------------------------------------------------------
>     SIP/2.0 100 Trying
>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     CSeq: 102 BYE
>     Timestamp: 1347610034 0.000000
>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>     Content-Length: 0
>
> ------------------------------------------------------------------------
> send 510 bytes to udp/[192.168.117.8]:5060 at 08:06:51.043021:
> ------------------------------------------------------------------------
>     SIP/2.0 480 Temporarily Unavailable
>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     CSeq: 102 BYE
>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>     Supported: timer, precondition, path, replaces
>     Content-Length: 0
>
> ------------------------------------------------------------------------
> recv 363 bytes from udp/[192.168.117.8]:5060 at 08:06:51.046021:
> ------------------------------------------------------------------------
>     SIP/2.0 200 OK
>     Via: SIP/2.0/UDP 192.168.117.9;rport;branch=z9hG4bKvBN2ZB9DZKZ6c
>     From: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>     To: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>     Server: Cisco-SIPGateway/IOS-12.x
>     Content-Length: 0
>     CSeq: 33468941 BYE
>
> ------------------------------------------------------------------------
>
> Thanks,
>
> Denis
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 djbinter at gmail.com  Fri Sep 14 19:32:23 2012
From: djbinter at gmail.com (Dorn DJBinter)
Date: Fri, 14 Sep 2012 08:32:23 -0700
Subject: [Freeswitch-users] T.38 with Re-INVITE question
In-Reply-To: <CAMwZ-t+f7dnfj3DakFxnfZTGi+J44Gawn3ycPi59Aktdi5uyAA@mail.gmail.com>
References: <CAAB3jWuXq67RX6j3=Zcv=9H432Ot=JkrrfMfrUVdi99Gf-VSPA@mail.gmail.com>
	<CAMwZ-t+f7dnfj3DakFxnfZTGi+J44Gawn3ycPi59Aktdi5uyAA@mail.gmail.com>
Message-ID: <-2854908417897542831@unknownmsgid>

Thank you for your suggestion; however, as mentioned, both CDRs are
a-leg and related to only the T.38 re-INVITE call behavior only.

Sent from my iPad


On Sep 14, 2012, at 2:38 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, Sep 13, 2012 at 5:07 PM, DJB International <djbinter at gmail.com> wrote:
>> I was testing T.38 with Re-INVITE.
>>
>> Call Flow:  Inbound Carrier (1.2.3.4) -> FS (Bypass Media) -> Outbound
>> Carrier that will send T.38 Re-INVITE (6.7.8.9).
>>
>> However, I noticed that FS generated 2 CDRs for the same call, which are: 1)
>> A-leg CDR from inbound carrier, and also 2) A-leg CDR from outbound carrier
>> that send T.38 Re-INVITE.
>>
>> I am wondering whether this is a normal behavior, or there is a way to
>> ignore the 2nd CDR?
>
>
> Does this help?
>
> process_cdr <-- indicates how to process CDR records.
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#process_cdr
>
>
> 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


From ssoni at lifesize.com  Fri Sep 14 15:56:35 2012
From: ssoni at lifesize.com (sanjay)
Date: Fri, 14 Sep 2012 04:56:35 -0700 (PDT)
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTikYfG9d=JXqWeG7nYF-miH-FmvpJVgS8X-DEJWF@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
	<AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
	<AANLkTikqtNE2AZEkPdD=cmf+_GV=Swr9yROTERytP+rC@mail.gmail.com>
	<AANLkTinSHvXi6sj9S4AO_zeqq+so=5P4LytB48q+H-f8@mail.gmail.com>
	<1297716217329-6025112.post@n2.nabble.com>
	<AANLkTikYfG9d=JXqWeG7nYF-miH-FmvpJVgS8X-DEJWF@mail.gmail.com>
Message-ID: <1347623795705-7582840.post@n2.nabble.com>

Was there any update after this reply of 'no' ? I dont see any JIRA# also
where I can find the solution. 
 please update this. I am getting similar issue where on a inbound call
gtalk not sending audio RTP packets. I will appreciate update on this. 
Thanks. 





--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-inbound-audio-tp6005151p7582840.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ssoni at lifesize.com  Fri Sep 14 16:07:08 2012
From: ssoni at lifesize.com (sanjay)
Date: Fri, 14 Sep 2012 05:07:08 -0700 (PDT)
Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no
	audio?
In-Reply-To: <33c87fa30906231941m14d56dear6958288a54057137@mail.gmail.com>
References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com>
	<33c87fa30906231941m14d56dear6958288a54057137@mail.gmail.com>
Message-ID: <1347624428589-7582841.post@n2.nabble.com>

Mark, Did you finally get the audio for your setup ? Can you please update
the solution ? I know its 3 long years! In my set-up I have reached till
this point. Both side STUN requests are being responded successfully. SIP
device -> FS -> Gtalk RTP packets are going but gtlak is not plying them and
it is also not sending any RTP to FS !!



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-picking-wrong-IP-address-no-audio-tp3142289p7582841.html
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From msc at freeswitch.org  Fri Sep 14 20:38:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Sep 2012 09:38:44 -0700
Subject: [Freeswitch-users] Altering From Header in SIP Invite
In-Reply-To: <49C5FCA19A8A114493EBAACA42FE5899105CA323@1AERDCEXCHMBX1.AER.AERCO.local>
References: <49C5FCA19A8A114493EBAACA42FE5899105C87C2@1AERDCEXCHMBX1.AER.AERCO.local>
	<49C5FCA19A8A114493EBAACA42FE5899105CA323@1AERDCEXCHMBX1.AER.AERCO.local>
Message-ID: <CAKzWOxUjc=LSQ8S5Hh6wd=ofNHg8-n=Uf2QebPNBkOwdNw_yJg@mail.gmail.com>

Without throwing the provider under the bus it might be good to document
this as a "known issue with a workaround." We would just need to add a
provider page and link it here:
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

Thanks!
-MC

On Fri, Sep 14, 2012 at 4:46 AM, Rob Moore <Rob.Moore at aeriandi.com> wrote:

>  Hi Everyone,****
>
> ** **
>
> For those who are interested I found a way around this issue. I?m not sure
> if this is the correct way to produce this result but it worked.****
>
> ** **
>
> The problem I had was that I could set the Freephone number I wished to
>  present but the P asserted ID would overwrite this with the other standard
> number I was attempting to send. ****
>
> ** **
>
> So to resolve the issue I had to disable all CID, use
> effective_caller_id_number to present my Freephone number (this adds the
> number to your from header) then I used sip_h_ to add the
> P-Asserted-Identity manually.****
>
> ** **
>
>                                 <action application="set"
> data="effective_caller_id_number=+448000000262"/>****
>
>                                 <action application="export"
> data="sip_cid_type=none"/>****
>
>                                 <action application="export"
> data="sip_h_P-Asserted-Identity= <sip:+445600000262 at fixsipbizzle.org> "/>*
> ***
>
> ** **
>
> Hope this helps anyone else who ever has to provide this unusual setup.***
> *
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rob Moore
> *Sent:* 11 September 2012 19:11
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Altering From Header in SIP Invite****
>
> ** **
>
> Hi All,****
>
> ** **
>
> I?m having a little trouble with ?presentation numbers? with a new
> provider I?m in IOT with this week.****
>
> ** **
>
> I?m trying to recreate the following Invite as the calls pass through our
> dialplan to this provider but there are issues with trying to get a
> different CLI into the From and P-Asserted-Identity headers.****
>
> ** **
>
> When presenting a Freephone number (for example) we need to still present
> the regular number that would be used by this extension in the
> P-Asserted-Identity whilst passing the number we wish to have presented in
> the From header.****
>
> ** **
>
> Currently we are not using Gateways so we cannot resort to using  <param
> name="caller-id-in-from" value="true"/> (although I expect this won?t do
> what we need in this case) so I?ve looked at altering the channel variables
>  sip_from_user,sip_full_from and sip_full_uri using set and export dial
> plan apps but none of these seem to have any effect so I guess these
> variables must be read only. ****
>
> ** **
>
> I?m sure this must be simple, but can?t for the life of me work out what I
> need to do. ****
>
> ** **
>
> Below is an extract from an example header from the provider I am trying
> to recreate, I?ve also added a copy of the Dialplan extension I am using to
> test.****
>
> If someone can tell me what I?m getting wrong I would really appreciate it.
> ****
>
> ** **
>
> Thanks****
>
> ** **
>
> Rob****
>
> ** **
>
> * *
>
> *INVITE* sip:*+445600005262 at primarysip.barfoo.com;user=phone* SIP/2.0****
>
> *Max-Forwards:* 69****
>
> *Session-Expires:* 3600;refresher=uac****
>
> *Min-SE:* 600****
>
> *Supported:* timer, 100rel****
>
> *To:* +445600000262 *<sip:+445600005262 at primarysip.barfoo.com;user=phone>*
> ****
>
> *From:* *<sip:+448000655262 at uk.foo.bar.2.net;user=phone>*;tag=12544****
>
> *P-Asserted-Identity:* <sip:+445600655262 at uk.sdin.bt.net>****
>
> *Call-ID:* 1347372978-13100 at mgc-uk-998.n2****
>
> *CSeq:* 1 INVITE****
>
> *Allow:* CANCEL, ACK, INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE****
>
> ** **
>
> ** **
>
> DIAL Plan: (attempting +445600655262 in P Asserted-Id and +448000655262in from) have commented out some things that I have tried.)
> ****
>
> If you are worried about [sip_h_X-Gateway=4003:4] this is added to help
> our SBC forward calls to our different providers.****
>
> ** **
>
>    <extension name="TC 25">****
>
>      <condition field="destination_number" expression="^10125$">****
>
>                                 <action application="set"
> data="effective_caller_id_name=_undef_"/>****
>
>                                 <action application="set"
> data="effective_caller_id_number=+445600005262 "/>****
>
>                                 <action application="set"
> data="dtmf_type=rfc2833"/>****
>
>                                 <action application="set"
> data="sip_cid_type=pid"/>****
>
>                                 <!--<action application="export" data="sip_from_user=
> +448000655262"/> -->****
>
>                                 <!-- <action application="export"
> data="sip_h_from=<sip:+448000655262 at primarysip.barfoo.com>;tag=${sip_from_tag}
> "/> -->****
>
>                                 <!--  <action application="export"
> data="sip_h_P-Asserted-Identity= +445600005262 <
> sip:+445600655262 at 172.17.7.4> "/> -->****
>
>                                 <action application="info"/>****
>
>                   <action application="bridge"
> data="[sip_h_X-Gateway=4003:4]sofia/internal/+442920001199 at pstn.tel" /> **
> **
>
>      </condition>****
>
>    </extension> ****
>
> ** **
>
> ** **
>
> Rob Moore****
>
> Telephony Systems Infrastructure Manager****
>
> *Aeriandi*
>
> Aeriandi Ltd, Prama House, Banbury Road, Oxford, OX27HT ****
>
> E: rob.moore at aeriandi.com**
>
> W: www.aeriandi.com
> M: +44 (0)7766 838040**
>
> T: +44 (0) 845 108 0308****
>
> ** **
>
> [image: Description: Description: Description: Description: Description:
> Description: Description: Description: cid:image002.png at 01CC9E0C.20153A40]
>      ****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 14 21:01:24 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Sep 2012 10:01:24 -0700
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
Message-ID: <CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>

Hi Peter!

Welcome to FreeSWITCH! Glad to have you. These are great questions. FS has
a db command that can be executed from fs_cli, however this isn't what you
want/need for users. The user directory is where all that stuff is handled.
Static XML is, of course, in conf/directory/* and you can add/remove there,
but that doesn't sound like what you need. The best option for you right
now is to get up to speed with mod_xml_curl. Using mod_xml_curl you can
have FS poll a database each time there is a need, such as when there is an
auth challenge taking place.

We have several people who've done lots of mod_xml_curl stuff, so feel free
to ask away, but please check the wiki first since there is a lot of
information and examples there. You may also enjoy discussing this live in
#freeswitch channel on irc.freenode.net.

-MC

On Fri, Sep 14, 2012 at 2:03 AM, Peter van Raamsdonk <peetzer at gmail.com>wrote:

> Hi all,
>
> Just wondering if I can add/ delete users to the db from the CLI (I'm
> using ESL java library)?
>
> See http://wiki.freeswitch.org/wiki/Function_db
>
> From the Dialplan this is possible, I want however to dynamically add new
> users.
>
> <action application="db" data="insert/realm/key/value"/>
>
>
> In the api (mod commands) nothing is mentioned about "db"
>
> I don't know if something like this could work with my example extension
> 1999, default group named default...
>
> response = client.sendSyncApiCommand( "db", "insert/default/1999/1999");
>
> Another option is to use mod_xml_curl and feed the dialplan or user
> directory to FreeSwitch back again.
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 14 21:07:24 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Sep 2012 10:07:24 -0700
Subject: [Freeswitch-users] calls remain connected after conference
 initiator hangs up
In-Reply-To: <DUB002-W1529765E14C38AEFFB295A8D9910@phx.gbl>
References: <DUB002-W1529765E14C38AEFFB295A8D9910@phx.gbl>
Message-ID: <CAKzWOxW31joLF3zSd0aG+5j8i3PTUn7AbXaD7s85UmZGWG63uA@mail.gmail.com>

I'm sure that this is all possible. First, though, pastebin your conference
configs and the relevant part of the dialplan that controls sending calls
into the conference. Also, pastebin a debug log of calls going into the
conference, including the moderator.

If you have not already done so, be sure to set the wait-mod flag for the
members going in and the moderator and endconf flags for the moderator
going in:
http://wiki.freeswitch.org/wiki/Mod_conference#Examples

-MC

On Thu, Sep 13, 2012 at 3:10 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com
> wrote:

> Hi, i have a little problem that i m sure it is easy to fix , my
> extensions make conferences (the confonference button on the phone)
> and/or transfers and then they hag up but the others in the conference
> remain
> connected, so this is provoking some people to abuse by connecting their
> friends
> through the company PBX and then they hang up and leave the friends
> connected in the call ... i have not succeeded so far in finding how to
> fix this.
>
> Also, it is possible to allow to transfer calls from outside to inside
> and/or inside to outside
> but not to transfer calls from outside to outside ?
> 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
>
>


-- 
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 Sep 14 21:13:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Sep 2012 10:13:25 -0700
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
Message-ID: <CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>

Hi Alex,

Welcome to the FreeSWITCH mail list!

First question for you: what kind of telephone are you using? The reason I
ask is that this kind of function is trivially achieved with a good hard
phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys.
If you can use a hard phone w/ multiple line keys then you don't even need
to mess with the dialplan, uuid_bridge, etc.

-MC

On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de>wrote:

>  Hi All,****
>
> ** **
>
> I?m new on the mailing list.****
>
> I have a problem with a call scenario.****
>
> **-          **Channel A and channel B are bridged (A is my own channel
> and B is my calling partner)****
>
> **-          **Now i set channel B on hold with the command ?uuid_hold
> xxx? and create a new channel to C with the command:****
>
> bgapi originate
> {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip
> ',origination_caller_id_number=num at ip}sofia/external/num at ip  &park****
>
> this works correctly, the partner C answer and the channel is established.
> ****
>
> **-          **Now the Problem:****
>
> I try to bridge the channel a to channel c with the uuid_bridge command,
> now the channel b will hangup, why? The variable hangup_after_bridge is by
> default false.****
>
> Other problem: channel A can hear the voice of channel C but not speak
> with him, channel C can hear and speak. But this problem is not the
> important think at the moment.****
>
> Is there a general problem in my plan to do that?****
>
> Is it a better plan to do this over the dialplan?****
>
> The next step in this scenario is to toggle the connection A -> B and A ->
> C.****
>
> ** **
>
> Thanks for your help!****
>
> Nice regards,****
>
> 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From robert.hadley at teotech.com  Fri Sep 14 22:06:08 2012
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 14 Sep 2012 18:06:08 +0000
Subject: [Freeswitch-users] Call screen for call to multiple endpoints not
 hanging up other endpoints on ANSWER
Message-ID: <71943DD5C22943448A24B7C5CDC238071E2B69FD@CH1PRD0411MB430.namprd04.prod.outlook.com>


I am using call screen feature and it works great for calls to single endpoints.  However when call bridges to multiple endpoints the other endpoints continue to ring after one endpoint is answered.  They are hungup on the accept/reject, but they should be hungup on answer.

These are the variables I am setting up before call_screen:
               switch_channel_set_variable(channel, "ignore_early_media", "true");
               switch_channel_set_variable(channel, "send_silence_when_idle", "10000");            // Eliminates initial white noise
               p = switch_channel_get_variable(channel, "caller_id_number");
               s = switch_channel_get_variable(channel, "uuid");
               switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "screen_call: caller_id_number='%s', variable_uuid='%s'\n", p, s);
               switch_snprintf(q, 256, "/tmp/%s-%s-name.wav", p, s);
               switch_channel_set_variable(channel, "call_screen_filename", q);
               switch_channel_set_variable(channel, "group_confirm_key", "exec");
               switch_core_session_execute_application(session, "answer", NULL);
               switch_core_session_execute_application(session, "sleep", "1000");
               switch_snprintf(r, 256, "callscreen_record_name,%s", dn);
               switch_core_session_execute_application(session, "phrase", r);
               switch_core_session_execute_application(session, "playback", "tone_stream://%(500, 0, 640)");
               switch_channel_set_variable(channel, "playback_terminators", "#*0123456789");
               switch_snprintf(r, 256, "%s 7 200 2", q);
               switch_core_session_execute_application(session, "record", r);
               switch_core_session_execute_application(session, "set", "ringback=${hold_music}");
               switch_core_session_execute_application(session, "set", "transfer_ringback=${hold_music}");
               switch_channel_set_variable(channel, "group_confirm_key", "exec");
               switch_channel_set_variable(channel, "group_confirm_cancel_timeout", "60");
               switch_channel_set_variable(channel, "origination_cancel_key", "2");
               switch_channel_set_variable(channel, "group_confirm_read_timeout", "60");
               switch_channel_set_variable(channel, "group_confirm_error_file", "/opt/teoswitch/lang/en/vm/vm-goodbye.wav");
               switch_channel_set_variable(channel, "fail_on_single_reject", "true");
               switch_channel_set_variable(channel, "continue_on_fail", "true");
               switch_channel_set_variable(channel, "call_timeout", "60");
               p = switch_channel_get_variable(channel, "uuid");
               switch_snprintf(r, 256, "callscreen %s %s", dn, p);
               switch_channel_set_variable(channel, "group_confirm_file", r);

Thanks,
Robert Hadley
Software Engineer

[Description: Description: Teo Logos (R)]
Teo, formerly Tone Commander Systems
UC            425.349.1045
WEB         www.teotech.com<http://www.teotech.com/>

CONNECT WITH US   Facebook<http://facebook.com/TeoUC>  |  Twitter<http://twitter.com/TeoTechUC>  |  YouTube<http://youtube.com/TeoMedia>


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From anthony.minessale at gmail.com  Fri Sep 14 23:08:35 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Sep 2012 14:08:35 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <088228C8-3C8F-455A-9280-4D7B371DC1C2@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
	<088228C8-3C8F-455A-9280-4D7B371DC1C2@me.com>
Message-ID: <CAKbxfG_yniysMEmqZYumtP2oR-GvYD_Y0RfYyesY4wYgy_kzcg@mail.gmail.com>

You did it again...

I need the DEBUG level in the logs

sofia global siptrace on
console loglevel debug
fsctl loglevel debug


On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> So what I am seeing is if the call flow is like
>
> Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
>
> but if the flow is
>
> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs
> --> PSTN things seem broken now if I remove wait for ACK from Server A's
> B-Leg the call works as expected
>
> FS_SERVER_A
> http://pastebin.freeswitch.org/19898
>
> FS_SERVER_B
> http://pastebin.freeswitch.org/19899
>
> On Sep 13, 2012, at 3:58 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> sigh, you forgot "console loglevel debug"  this is only info
> loglevel.........
>
>
> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> So it seems all is not well - missing ACK and forward of 200OK when 2 FS
>> servers are in the mix when sip_wait_for_aleg_ack=true is enabled
>> basic flow is
>>
>>                         FS Server A
>>                FS Server B
>> |-----------------DID Channel---------------------|
>> |---------Termination Channel (RCF)----------|
>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs
>> --> Carrier
>>
>> Both FS running latest GIT as of this morning - Both FS servers
>> using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not
>> getting an ACK back from FS_A)
>>
>> FS Server A Logs
>> http://pastebin.freeswitch.org/19897
>>
>> FS Server B SIP Logs
>> http://pastebin.freeswitch.org/19896
>>
>>
>>
>>
>>
>>
>> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>>
>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com
>> > wrote:
>>
>>> Crash was resolved this weekend by a code update via jira case - no more
>>> crashes after the update - some more dial plan edits and a little more
>>> testing I will be ready to head over to pay the Wiki tax
>>>
>>> Thanks guys for adding this feature it seems to be working as expected
>>> and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
>>>
>>
>> Thanks for checking in and letting us know where you stand. I really like
>> it when these threads have some sort of resolution. If you have any issues
>> with wiki editing please let me know or hop on IRC for an assist.
>>
>> -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
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jpablolorenzetti at hotmail.com  Fri Sep 14 23:28:57 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 14 Sep 2012 19:28:57 +0000
Subject: [Freeswitch-users] calls remain connected after conference
 initiator hangs up
In-Reply-To: <CAKzWOxW31joLF3zSd0aG+5j8i3PTUn7AbXaD7s85UmZGWG63uA@mail.gmail.com>
References: <DUB002-W1529765E14C38AEFFB295A8D9910@phx.gbl>,
	<CAKzWOxW31joLF3zSd0aG+5j8i3PTUn7AbXaD7s85UmZGWG63uA@mail.gmail.com>
Message-ID: <DUB002-W110B62E4BFA5A29A9E7DDC9D9900@phx.gbl>

Hi, thanks for the response, i did not explain my self good i think, it is not a conference it is a 3 way call (on the phone the buttons says "conference", sorry for the misunderstanding). what is happening is the following scenario:
for the 3 way call:
Internal extension calls out through the gw to friend A, puts friend A on hold and calls out through the gw to friend B. the the 3 of them are connected and can talk. the internal extension then hangs the phone but friend A and friend B remain connected.
for the transfers:i would like to stop transfers in which an internal extension transfers a call (not originated within my PBX but came from somewhere else like a gw) to outside lines. what is happenning is that some people are contacting their friends who work for the company and ask them to transfer them to some other number in different districts, so they make a call transfer using the extension and then hang and the call remains connected. these people pay for the local portion of the call and the company is paying for the national long distance portion of the call. again people are abusing this.
both cases are actually the same problem just different environments.i hope this time is clear.
thanks!Date: Fri, 14 Sep 2012 10:07:24 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] calls remain connected after conference initiator hangs up

I'm sure that this is all possible. First, though, pastebin your conference configs and the relevant part of the dialplan that controls sending calls into the conference. Also, pastebin a debug log of calls going into the conference, including the moderator.


If you have not already done so, be sure to set the wait-mod flag for the members going in and the moderator and endconf flags for the moderator going in:
http://wiki.freeswitch.org/wiki/Mod_conference#Examples


-MC

On Thu, Sep 13, 2012 at 3:10 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:




Hi, i have a little problem that i m sure it is easy to fix , my extensions make conferences (the confonference button on the phone) and/or transfers and then they hag up but the others in the conference remain
connected, so this is provoking some people to abuse by connecting their friends through the company PBX and then they hang up and leave the friendsconnected in the call ... i have not succeeded so far in finding how to fix this. 

Also, it is possible to allow to transfer calls from outside to inside and/or inside to outsidebut not to transfer calls from outside to outside ?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




-- 
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 peetzer at gmail.com  Fri Sep 14 21:54:28 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Fri, 14 Sep 2012 19:54:28 +0200
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
Message-ID: <C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>

Hi Michael and group,

I tried the mod_xml_curl after studying the wiki. It works like a charm and FS post to my java servlet easily (dialplan and user directory).
I read on a blog there is option 'cacheable=true' to prevent a post every time a dial is made, do you know where to put this?

I suspected that the reloadxml would trigger a post but it doesn't, dialplan is triggered from dial and user directory from startup of FS and dial I thought.

FS is highly customizable and I need to find my way but it works like a charm.

I wonder if there are people who had a successful mod_xml_java build on Windows. I made a own project and put references to the core but got stuck and left it because it took too much time.

Thanks and have a nice weekend all!

Peter

On 14 sep 2012, at 19:01, Michael Collins wrote:

> Hi Peter!
> 
> Welcome to FreeSWITCH! Glad to have you. These are great questions. FS has a db command that can be executed from fs_cli, however this isn't what you want/need for users. The user directory is where all that stuff is handled. Static XML is, of course, in conf/directory/* and you can add/remove there, but that doesn't sound like what you need. The best option for you right now is to get up to speed with mod_xml_curl. Using mod_xml_curl you can have FS poll a database each time there is a need, such as when there is an auth challenge taking place.
> 
> We have several people who've done lots of mod_xml_curl stuff, so feel free to ask away, but please check the wiki first since there is a lot of information and examples there. You may also enjoy discussing this live in #freeswitch channel on irc.freenode.net.
> 
> -MC
> 
> On Fri, Sep 14, 2012 at 2:03 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
> Hi all,
> 
> Just wondering if I can add/ delete users to the db from the CLI (I'm using ESL java library)?
> 
> See http://wiki.freeswitch.org/wiki/Function_db
> 
> From the Dialplan this is possible, I want however to dynamically add new users.
> 
> <action application="db" data="insert/realm/key/value"/>
> 
> In the api (mod commands) nothing is mentioned about "db"
> 
> I don't know if something like this could work with my example extension 1999, default group named default...
> 
> response = client.sendSyncApiCommand( "db", "insert/default/1999/1999");
> 
> Another option is to use mod_xml_curl and feed the dialplan or user directory to FreeSwitch back again.
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike.burlingame at me.com  Fri Sep 14 23:42:10 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 12:42:10 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG_yniysMEmqZYumtP2oR-GvYD_Y0RfYyesY4wYgy_kzcg@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
	<088228C8-3C8F-455A-9280-4D7B371DC1C2@me.com>
	<CAKbxfG_yniysMEmqZYumtP2oR-GvYD_Y0RfYyesY4wYgy_kzcg@mail.gmail.com>
Message-ID: <5555D975-3334-48A6-815B-4367640429F5@me.com>

Sorry about that yesterday was a long day 

On Sep 14, 2012, at 12:08 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> You did it again...
> 
> I need the DEBUG level in the logs 
> 
> sofia global siptrace on
> console loglevel debug
> fsctl loglevel debug
> 
> 
> On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> So what I am seeing is if the call flow is like
> 
> Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
> 
> but if the flow is 
> 
> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected 
> 
> FS_SERVER_A 
> http://pastebin.freeswitch.org/19898
> 
> FS_SERVER_B
> http://pastebin.freeswitch.org/19899
> 
> On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> sigh, you forgot "console loglevel debug"  this is only info loglevel.........
>> 
>> 
>> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled  
>> basic flow is
>>  
>>                         FS Server A                                                      FS Server B
>> |-----------------DID Channel---------------------|     |---------Termination Channel (RCF)----------|
>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
>> 
>> Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
>> 
>> FS Server A Logs
>> http://pastebin.freeswitch.org/19897
>> 
>> FS Server B SIP Logs
>> http://pastebin.freeswitch.org/19896
>> 
>> 
>> 
>> 
>> 
>> 
>> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> 
>>> 
>>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
>>> 
>>> Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared 
>>> 
>>> Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
>>> 
>>> -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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gavin.henry at gmail.com  Sat Sep 15 00:30:11 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 14 Sep 2012 21:30:11 +0100
Subject: [Freeswitch-users] List of "status" statuses in a PRESENCE_IN event?
Message-ID: <CAA8_NKD0ZPvn-bRWzPxcd240V1-QbpvXNhVVLcsQDCbBiPXFgw@mail.gmail.com>

Hi all,

Given (/event plain PRESENCE_IN) :

RECV EVENT
Event-Name: PRESENCE_IN
Core-UUID: 68101dee-d889-4f0f-934c-3c2da2dae90c
FreeSWITCH-Hostname: test.co.uk
FreeSWITCH-Switchname: test.co.uk
FreeSWITCH-IPv4: test_ip
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2012-09-14 21:05:29
Event-Date-GMT: Fri, 14 Sep 2012 20:05:29 GMT
Event-Date-Timestamp: 1347653129346455
Event-Calling-File: sofia_reg.c
Event-Calling-Function: sofia_reg_del_callback
Event-Calling-Line-Number: 629
Event-Sequence: 9373756
proto: sip
rpid: away
login: sip:mod_sofia at test_ip:5060
user-agent: Registered(UDP)
from: 1688 at test.co.uk
status: Unregistered
event_type: presence


Is there a list of what the "status" can be? Above has user-agent:
Registered(UDP) but status: Unregistered. Is status if someone is
subscribed to this presence info, like a BLF?

How can I tell if a phone is online via a PRESENCE_IN?

Re the "status" values, I've seen:

CS_ROUTING == Ringing
CS_HANGUP == HangUp :-)
answered == Answered :-)

I've also seen:

status: Registered(TCP-NAT)

status: Unregistered but user-agent saying Registered(TCP-NAT) too.

How can I check the endpoint is available, i.e. which field and value
drives a BLF being green, red and flashing? (I know that's obviously
done via a SIP NOTIFY)

I'm trying to push out event messages via our API
(http://www.surevoip.co.uk/api) showing "available", "busy" and
"ringing" like a BLF so you can register for a WebHook.

We're currently sponsoring (via FreeSWITCH consulting point of
contact) Presence via XMPP as we run our own XMPP servers too. That
info will get documented here:

http://wiki.freeswitch.org/wiki/Presence#XMPP_presence

but I want to expose this via our WebHooks too for HTTP POSTs.

Thanks,

Gavin.

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


From paul at cupis.co.uk  Sat Sep 15 00:49:36 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Fri, 14 Sep 2012 21:49:36 +0100
Subject: [Freeswitch-users] Altering From Header in SIP Invite
In-Reply-To: <49C5FCA19A8A114493EBAACA42FE5899105CA323@1AERDCEXCHMBX1.AER.AERCO.local>
References: <49C5FCA19A8A114493EBAACA42FE5899105C87C2@1AERDCEXCHMBX1.AER.AERCO.local>
	<49C5FCA19A8A114493EBAACA42FE5899105CA323@1AERDCEXCHMBX1.AER.AERCO.local>
Message-ID: <50539860.7070302@cupis.co.uk>

On 14/09/12 12:46, Rob Moore wrote:
> Hope this helps anyone else who ever has to provide this unusual setup.

This does seem to be the only way of dealing with this setup.

You may additionally want to look at doing the following, depending on 
the rest of your setup.

Remove P-A-ID headers before adding custom header:
     <action application="unset" data="sip_h_P-Asserted-Identity"/>

Honour Privacy/withheld-flag:
     <action application="unset" data="sip_h_Privacy"/>
     <action application="set" data="sip_h_Privacy=id"/>

(some might also want to set Privacy:none explicitly when CLI may be 
released, but the absence of any Privacy flags should set this by default).

Regards,


From lists at kavun.ch  Sat Sep 15 00:50:22 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 14 Sep 2012 16:50:22 -0400
Subject: [Freeswitch-users] 481 Call Leg/Transaction Does Not Exist
In-Reply-To: <5D5FD0B1-34C0-4C06-AD20-735E3ACE8803@kavun.ch>
References: <5D5FD0B1-34C0-4C06-AD20-735E3ACE8803@kavun.ch>
Message-ID: <99D9E8DC-8621-47E4-8F25-44F7B33001B2@kavun.ch>

Hi all,

I have tried to play with the refer extension and now have my calls land in the public context when they are transferred?
This is the second extension in my dialplan:
<extension name="refer">
 <condition field="${sip_refer_to}">
        <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression>
 </condition>
 <condition field="${sip_refer_to}">
        <expression><![CDATA[<sip:(.*)@(.*)>]]></expression>
        <action application="set" data="refer_user=$1"/>
        <action application="set" data="refer_domain=$2"/>
        <action application="info"/>
        <action application="bridge" data="sofia/${use_profile}/${refer_user}@${domain_name}"/>
 </condition>
    </extension>

Any help would be greatly appreciated.

All the best,
Emrah
On Sep 13, 2012, at 3:09 PM, Emrah <lists at kavun.ch> wrote:

> Hi all,
> 
> I seem to be encountering the strangest problems.
> I have been doing some more testing with my transfers and here is a recurring issue.
> 
> I have 2 phones behind the same router. 1 Polycom VVX 1500 and 1 Yealink SIP-T38g
> They are both registering on the same FS server which is on a remote public IP.
> The Polycom has extension 20 and the yealink uses 21.
> When I call the Polycom from the Yealink and perform a blind transfer from the Yealink, it fails with: 481 Call Leg/Transaction Does Not Exist.
> If I transfer from the Polycom though, it seems to work.
> 
> In resume:
> I cannot transfer from A leg. B leg doesn't seem to know its way around and looses it. To demonstrate, I transfer B leg from A leg to extension 5000.
> http://pastebin.freeswitch.org/19893
> 
> I tried Googleing the error code and didn't find anything particularly related to my issue.
> 
> Any hint would be greatly appreciated. To make sure, I downgraded to the stable version.
> 
> Thanks a bunch, as always.
> Emrah



From anthony.minessale at gmail.com  Sat Sep 15 01:29:26 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Sep 2012 16:29:26 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <5555D975-3334-48A6-815B-4367640429F5@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
	<088228C8-3C8F-455A-9280-4D7B371DC1C2@me.com>
	<CAKbxfG_yniysMEmqZYumtP2oR-GvYD_Y0RfYyesY4wYgy_kzcg@mail.gmail.com>
	<5555D975-3334-48A6-815B-4367640429F5@me.com>
Message-ID: <CAKbxfG87nwbn+BHip_RoP87A=+paw9tbLKdw-AhMYLiMan4DkA@mail.gmail.com>

It was getting tripped up from the bypass media invite.  Try latest.


On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> Sorry about that yesterday was a long day
>
> On Sep 14, 2012, at 12:08 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> You did it again...
>
> I need the DEBUG level in the logs
>
> sofia global siptrace on
> console loglevel debug
> fsctl loglevel debug
>
>
> On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> So what I am seeing is if the call flow is like
>>
>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
>>
>> but if the flow is
>>
>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs
>> --> PSTN things seem broken now if I remove wait for ACK from Server A's
>> B-Leg the call works as expected
>>
>> FS_SERVER_A
>> http://pastebin.freeswitch.org/19898
>>
>> FS_SERVER_B
>> http://pastebin.freeswitch.org/19899
>>
>> On Sep 13, 2012, at 3:58 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> sigh, you forgot "console loglevel debug"  this is only info
>> loglevel.........
>>
>>
>> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>>
>>> So it seems all is not well - missing ACK and forward of 200OK when 2 FS
>>> servers are in the mix when sip_wait_for_aleg_ack=true is enabled
>>> basic flow is
>>>
>>>                         FS Server A
>>>                  FS Server B
>>> |-----------------DID Channel---------------------|
>>> |---------Termination Channel (RCF)----------|
>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS -->
>>> OpenSIPs --> Carrier
>>>
>>> Both FS running latest GIT as of this morning - Both FS servers
>>> using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not
>>> getting an ACK back from FS_A)
>>>
>>> FS Server A Logs
>>> http://pastebin.freeswitch.org/19897
>>>
>>> FS Server B SIP Logs
>>> http://pastebin.freeswitch.org/19896
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>
>>>
>>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <
>>> mike.burlingame at me.com> wrote:
>>>
>>>> Crash was resolved this weekend by a code update via jira case - no
>>>> more crashes after the update - some more dial plan edits and a little more
>>>> testing I will be ready to head over to pay the Wiki tax
>>>>
>>>> Thanks guys for adding this feature it seems to be working as expected
>>>> and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
>>>>
>>>
>>> Thanks for checking in and letting us know where you stand. I really
>>> like it when these threads have some sort of resolution. If you have any
>>> issues with wiki editing please let me know or hop on IRC for an assist.
>>>
>>> -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
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
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From lists at kavun.ch  Sat Sep 15 01:30:33 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 14 Sep 2012 17:30:33 -0400
Subject: [Freeswitch-users] You don't need to save a greeting to override
	your existing one in Voicemail
Message-ID: <F07A307C-BE0A-43C7-8C88-0B7C411C7D8E@kavun.ch>

It used to be the case with Asterisk as well, now it has changed.
Basically, if you have a greeting in place and want to record a new one? And you suddenly cough in the middle of your recording? Than get interrupted by your boss right after you've pressed #? You just hang up the phone and think that your existing greeting hasn't been altered because you didn't press 2 to save your re-recorded greeting?
Well that's wrong.

Your existing greeting gets overridden as soon as you start the recording. Even if you hang up the phone in the middle of your message, it'll still be recorded and played as your greeting.

I don't think that this calls for a workaround or more studying of the app. I am using the default settings on this and think that it can be improved, so that you can review your greeting before committing it with a save action.
Being able to listen to your existing greeting before re-recording it wouldn't be a bad add-on either.

What is your take on this?

Emrah

From gabe at gundy.org  Sat Sep 15 01:33:16 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 14 Sep 2012 15:33:16 -0600
Subject: [Freeswitch-users] T.38 with Re-INVITE question
In-Reply-To: <-2854908417897542831@unknownmsgid>
References: <CAAB3jWuXq67RX6j3=Zcv=9H432Ot=JkrrfMfrUVdi99Gf-VSPA@mail.gmail.com>
	<CAMwZ-t+f7dnfj3DakFxnfZTGi+J44Gawn3ycPi59Aktdi5uyAA@mail.gmail.com>
	<-2854908417897542831@unknownmsgid>
Message-ID: <CAMwZ-t+QiAH7J58uY_b4FcGmZf_T0FG+02DVo9q3Yesm-2MjnQ@mail.gmail.com>

On Fri, Sep 14, 2012 at 9:32 AM, Dorn DJBinter <djbinter at gmail.com> wrote:
> Thank you for your suggestion; however, as mentioned, both CDRs are
> a-leg and related to only the T.38 re-INVITE call behavior only.

I should have seen that; it rather *was* late when I read it :)

As far as I know, you're going to have to insert a variable based on
your logic to tell FreeSWITCH that it should or shouldn't log.

Sorry I wasn't more helpful.


Gabe


From mike.burlingame at me.com  Sat Sep 15 01:38:05 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 14:38:05 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG87nwbn+BHip_RoP87A=+paw9tbLKdw-AhMYLiMan4DkA@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>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
	<088228C8-3C8F-455A-9280-4D7B371DC1C2@me.com>
	<CAKbxfG_yniysMEmqZYumtP2oR-GvYD_Y0RfYyesY4wYgy_kzcg@mail.gmail.com>
	<5555D975-3334-48A6-815B-4367640429F5@me.com>
	<CAKbxfG87nwbn+BHip_RoP87A=+paw9tbLKdw-AhMYLiMan4DkA@mail.gmail.com>
Message-ID: <FCCD44D4-0682-4669-BA9C-E370BD99584C@me.com>

Thanks updating and will report back

On Sep 14, 2012, at 2:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> It was getting tripped up from the bypass media invite.  Try latest.
> 
> 
> On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> Sorry about that yesterday was a long day 
> 
> On Sep 14, 2012, at 12:08 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> You did it again...
>> 
>> I need the DEBUG level in the logs 
>> 
>> sofia global siptrace on
>> console loglevel debug
>> fsctl loglevel debug
>> 
>> 
>> On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> So what I am seeing is if the call flow is like
>> 
>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
>> 
>> but if the flow is 
>> 
>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected 
>> 
>> FS_SERVER_A 
>> http://pastebin.freeswitch.org/19898
>> 
>> FS_SERVER_B
>> http://pastebin.freeswitch.org/19899
>> 
>> On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> sigh, you forgot "console loglevel debug"  this is only info loglevel.........
>>> 
>>> 
>>> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled  
>>> basic flow is
>>>  
>>>                         FS Server A                                                      FS Server B
>>> |-----------------DID Channel---------------------|     |---------Termination Channel (RCF)----------|
>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
>>> 
>>> Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
>>> 
>>> FS Server A Logs
>>> http://pastebin.freeswitch.org/19897
>>> 
>>> FS Server B SIP Logs
>>> http://pastebin.freeswitch.org/19896
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> 
>>>> 
>>>> 
>>>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
>>>> 
>>>> Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared 
>>>> 
>>>> Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
>>>> 
>>>> -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
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 mike.burlingame at me.com  Sat Sep 15 01:42:53 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 14:42:53 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <FCCD44D4-0682-4669-BA9C-E370BD99584C@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
	<088228C8-3C8F-455A-9280-4D7B371DC1C2@me.com>
	<CAKbxfG_yniysMEmqZYumtP2oR-GvYD_Y0RfYyesY4wYgy_kzcg@mail.gmail.com>
	<5555D975-3334-48A6-815B-4367640429F5@me.com>
	<CAKbxfG87nwbn+BHip_RoP87A=+paw9tbLKdw-AhMYLiMan4DkA@mail.gmail.com>
	<FCCD44D4-0682-4669-BA9C-E370BD99584C@me.com>
Message-ID: <A7B679D0-9AE2-45E2-99DF-720FE8BE49F9@me.com>

That addressed that failure - I am testing a few other failures and will report back thanks for the quick updates

On Sep 14, 2012, at 2:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:

> Thanks updating and will report back
> 
> On Sep 14, 2012, at 2:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> It was getting tripped up from the bypass media invite.  Try latest.
>> 
>> 
>> On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> Sorry about that yesterday was a long day 
>> 
>> On Sep 14, 2012, at 12:08 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> You did it again...
>>> 
>>> I need the DEBUG level in the logs 
>>> 
>>> sofia global siptrace on
>>> console loglevel debug
>>> fsctl loglevel debug
>>> 
>>> 
>>> On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> So what I am seeing is if the call flow is like
>>> 
>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
>>> 
>>> but if the flow is 
>>> 
>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected 
>>> 
>>> FS_SERVER_A 
>>> http://pastebin.freeswitch.org/19898
>>> 
>>> FS_SERVER_B
>>> http://pastebin.freeswitch.org/19899
>>> 
>>> On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> 
>>>> sigh, you forgot "console loglevel debug"  this is only info loglevel.........
>>>> 
>>>> 
>>>> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled  
>>>> basic flow is
>>>>  
>>>>                         FS Server A                                                      FS Server B
>>>> |-----------------DID Channel---------------------|     |---------Termination Channel (RCF)----------|
>>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
>>>> 
>>>> Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
>>>> 
>>>> FS Server A Logs
>>>> http://pastebin.freeswitch.org/19897
>>>> 
>>>> FS Server B SIP Logs
>>>> http://pastebin.freeswitch.org/19896
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>> 
>>>>> 
>>>>> 
>>>>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>> Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
>>>>> 
>>>>> Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared 
>>>>> 
>>>>> Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
>>>>> 
>>>>> -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
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From djbinter at gmail.com  Sat Sep 15 01:43:25 2012
From: djbinter at gmail.com (DJB International)
Date: Fri, 14 Sep 2012 14:43:25 -0700
Subject: [Freeswitch-users] T.38 with Re-INVITE question
In-Reply-To: <CAMwZ-t+QiAH7J58uY_b4FcGmZf_T0FG+02DVo9q3Yesm-2MjnQ@mail.gmail.com>
References: <CAAB3jWuXq67RX6j3=Zcv=9H432Ot=JkrrfMfrUVdi99Gf-VSPA@mail.gmail.com>
	<CAMwZ-t+f7dnfj3DakFxnfZTGi+J44Gawn3ycPi59Aktdi5uyAA@mail.gmail.com>
	<-2854908417897542831@unknownmsgid>
	<CAMwZ-t+QiAH7J58uY_b4FcGmZf_T0FG+02DVo9q3Yesm-2MjnQ@mail.gmail.com>
Message-ID: <CAAB3jWuzadUZpqgpfLq9hruinZhbUTZ9_qPogxEscG9sVnnSYA@mail.gmail.com>

Thank you.  I've already opened Jira for a bug report since it only happens
on latest git, but not on older version.

-djbinter

On Fri, Sep 14, 2012 at 2:33 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Sep 14, 2012 at 9:32 AM, Dorn DJBinter <djbinter at gmail.com> wrote:
> > Thank you for your suggestion; however, as mentioned, both CDRs are
> > a-leg and related to only the T.38 re-INVITE call behavior only.
>
> I should have seen that; it rather *was* late when I read it :)
>
> As far as I know, you're going to have to insert a variable based on
> your logic to tell FreeSWITCH that it should or shouldn't log.
>
> Sorry I wasn't more helpful.
>
>
> 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 anthony.minessale at gmail.com  Sat Sep 15 02:09:25 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Sep 2012 17:09:25 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <A7B679D0-9AE2-45E2-99DF-720FE8BE49F9@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>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
	<5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>
	<CAKzWOxVT+=bLMXMXe1fgsv4Vn8JFw8gRDF8xypJ-14KSVaKQAw@mail.gmail.com>
	<071E375F-B14D-46DA-899F-38F3725F0C92@me.com>
	<1D1B9033-340F-469C-8A79-D33A79071BFF@me.com>
	<CAKbxfG8TXVieh4gW5epyQSrh3Qvv8tUUiZ1oO3CC-HpFfE_=9A@mail.gmail.com>
	<283776BC-0455-4C16-8904-13396619CA3A@me.com>
	<CAKbxfG-qXCWiqiyEO5HSww03erZE3F48SFAyr71ie6WtJMEFmA@mail.gmail.com>
	<927793A3-06D4-4AFD-8AEC-69745E2BC775@me.com>
	<FFB50BF7-5D27-4DC8-A8DE-E6F5FF4585D4@me.com>
	<0FF83E36-3D20-4AAF-9D42-D841A7DCBEAC@me.com>
	<CAKbxfG9HGrr4joJe=5kC+BbU9PucFQjQNNn2UXdyv=9KG+TT=g@mail.gmail.com>
	<C8EE1C15-B10E-47DA-9A6C-6F682B7E551B@me.com>
	<C640A2A2-7548-448F-8DCF-C0918289B09B@me.com>
	<9482D8B7-1B61-4EA9-BA45-9473699D1D74@me.com>
	<CAKzWOxWOHwmUs=CoDjONq_Eg+wgFGiZfJtUKB1HUBU2p7fBPrQ@mail.gmail.com>
	<64F1449C-A35E-48E0-88F6-595C18F7D0DC@me.com>
	<CAKbxfG-xp=5pOz0ykhHwgDByu4imxqdJLm4y9KqWfcD1Q0hqHw@mail.gmail.com>
	<088228C8-3C8F-455A-9280-4D7B371DC1C2@me.com>
	<CAKbxfG_yniysMEmqZYumtP2oR-GvYD_Y0RfYyesY4wYgy_kzcg@mail.gmail.com>
	<5555D975-3334-48A6-815B-4367640429F5@me.com>
	<CAKbxfG87nwbn+BHip_RoP87A=+paw9tbLKdw-AhMYLiMan4DkA@mail.gmail.com>
	<FCCD44D4-0682-4669-BA9C-E370BD99584C@me.com>
	<A7B679D0-9AE2-45E2-99DF-720FE8BE49F9@me.com>
Message-ID: <CAKbxfG8eS6AzmXKkPs_tf7Ywu_X_36nfe6mtw=irq-JruCaf9Q@mail.gmail.com>

Please report issues to jira.freeswitch.org and be aware we
have commercial support as explained in most of the footers on this mL.


On Fri, Sep 14, 2012 at 4:42 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> That addressed that failure - I am testing a few other failures and will
> report back thanks for the quick updates
>
> On Sep 14, 2012, at 2:38 PM, Mike Burlingame <mike.burlingame at me.com>
> wrote:
>
> Thanks updating and will report back
>
> On Sep 14, 2012, at 2:29 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> It was getting tripped up from the bypass media invite.  Try latest.
>
>
> On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> Sorry about that yesterday was a long day
>>
>> On Sep 14, 2012, at 12:08 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> You did it again...
>>
>> I need the DEBUG level in the logs
>>
>> sofia global siptrace on
>> console loglevel debug
>> fsctl loglevel debug
>>
>>
>> On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>>
>>> So what I am seeing is if the call flow is like
>>>
>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
>>>
>>> but if the flow is
>>>
>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS -->
>>> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from
>>> Server A's B-Leg the call works as expected
>>>
>>> FS_SERVER_A
>>> http://pastebin.freeswitch.org/19898
>>>
>>> FS_SERVER_B
>>> http://pastebin.freeswitch.org/19899
>>>
>>> On Sep 13, 2012, at 3:58 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>> sigh, you forgot "console loglevel debug"  this is only info
>>> loglevel.........
>>>
>>>
>>> On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame at me.com
>>> > wrote:
>>>
>>>> So it seems all is not well - missing ACK and forward of 200OK when 2
>>>> FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled
>>>> basic flow is
>>>>
>>>>                         FS Server A
>>>>                  FS Server B
>>>> |-----------------DID Channel---------------------|
>>>> |---------Termination Channel (RCF)----------|
>>>> Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS -->
>>>> OpenSIPs --> Carrier
>>>>
>>>> Both FS running latest GIT as of this morning - Both FS servers
>>>> using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not
>>>> getting an ACK back from FS_A)
>>>>
>>>> FS Server A Logs
>>>> http://pastebin.freeswitch.org/19897
>>>>
>>>> FS Server B SIP Logs
>>>> http://pastebin.freeswitch.org/19896
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sep 11, 2012, at 4:56 PM, Michael Collins <msc at freeswitch.org>
>>>> wrote:
>>>>
>>>>
>>>>
>>>> On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <
>>>> mike.burlingame at me.com> wrote:
>>>>
>>>>> Crash was resolved this weekend by a code update via jira case - no
>>>>> more crashes after the update - some more dial plan edits and a little more
>>>>> testing I will be ready to head over to pay the Wiki tax
>>>>>
>>>>> Thanks guys for adding this feature it seems to be working as expected
>>>>> and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
>>>>>
>>>>
>>>> Thanks for checking in and letting us know where you stand. I really
>>>> like it when these threads have some sort of resolution. If you have any
>>>> issues with wiki editing please let me know or hop on IRC for an assist.
>>>>
>>>> -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
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
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From mike.burlingame at me.com  Sat Sep 15 02:10:42 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 15:10:42 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
Message-ID: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>

it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"

The ULC are stating we need to only have it in our supported and do not pass them a require.

Re-Invite from B-Leg to FS
   ------------------------------------------------------------------------
   INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
   Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
   Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
   Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
   Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
   Content-Length: 217
   Content-Type: application/sdp
   CSeq: 33480808 INVITE
   From: <sip:18475551212 at B-LEG_IP>;tag=100052073
   Max-Forwards: 92
   Session-Expires: 3600;refresher=uas
   Supported: timer
   To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
   User-Agent: FreeSwitch
   
   v=0
   o=- 3308986892 0 IN IP4 B-LEG_IP
   s=Media Server
   c=IN IP4 B-LEG_IP
   t=0 0
   m=audio 51246 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   ------------------------------------------------------------------------

Re-Invite from FS to A-Leg
  ------------------------------------------------------------------------
   INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
   Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
   Max-Forwards: 97
   From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
   To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
   Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
   CSeq: 33480811 INVITE
   Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
   User-Agent: FreeSwitch
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
   Require: timer
   Supported: timer, precondition, path, replaces
   Session-Expires: 64800;refresher=uas
   Min-SE: 64800
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 235
   X-FS-Support: update_display,send_info
   
   v=0
   o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
   s=Media Server
   c=IN IP4 B-LEG_IP
   t=0 0
   m=audio 51246 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   ------------------------------------------------------------------------



From anthony.minessale at gmail.com  Sat Sep 15 02:16:26 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Sep 2012 17:16:26 -0500
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
Message-ID: <CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>

set {sip_require_timer=false} in your outbound calls or globally


On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> it seems if I get an re-invite from the B-Leg FS add's requires timer and
> changes the session timer to a high value to the re-invite going to the
> A-Leg come to find out Acme Packets at our ULC's do not like this and send
> us back a 420 Bad Extension and the call is disconnected with Reason:
> Q.850;cause=127;text="INTERWORKING"
>
> The ULC are stating we need to only have it in our supported and do not
> pass them a require.
>
> Re-Invite from B-Leg to FS
>    ------------------------------------------------------------------------
>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>    Via: SIP/2.0/UDP
> B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>    Content-Length: 217
>    Content-Type: application/sdp
>    CSeq: 33480808 INVITE
>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>    Max-Forwards: 92
>    Session-Expires: 3600;refresher=uas
>    Supported: timer
>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>    User-Agent: FreeSwitch
>
>    v=0
>    o=- 3308986892 0 IN IP4 B-LEG_IP
>    s=Media Server
>    c=IN IP4 B-LEG_IP
>    t=0 0
>    m=audio 51246 RTP/AVP 0 101
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:20
>    ------------------------------------------------------------------------
>
> Re-Invite from FS to A-Leg
>   ------------------------------------------------------------------------
>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>    Max-Forwards: 97
>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy
> :5060;user=phone>;tag=c9Favaa53XFXB
>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER
> :5060;user=phone>;tag=SDd626401-gK095bbb72
>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>    CSeq: 33480811 INVITE
>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER
> :5070;transport=udp>
>    User-Agent: FreeSwitch
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>    Require: timer
>    Supported: timer, precondition, path, replaces
>    Session-Expires: 64800;refresher=uas
>    Min-SE: 64800
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 235
>    X-FS-Support: update_display,send_info
>
>    v=0
>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>    s=Media Server
>    c=IN IP4 B-LEG_IP
>    t=0 0
>    m=audio 51246 RTP/AVP 0 101
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:20
>    ------------------------------------------------------------------------
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike.burlingame at me.com  Sat Sep 15 02:38:35 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 15:38:35 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
Message-ID: <80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>

I have that set in the bridge command however does not seem to be working as expected.

add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");


On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> set {sip_require_timer=false} in your outbound calls or globally 
> 
> 
> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
> 
> The ULC are stating we need to only have it in our supported and do not pass them a require.
> 
> Re-Invite from B-Leg to FS
>    ------------------------------------------------------------------------
>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>    Content-Length: 217
>    Content-Type: application/sdp
>    CSeq: 33480808 INVITE
>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>    Max-Forwards: 92
>    Session-Expires: 3600;refresher=uas
>    Supported: timer
>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>    User-Agent: FreeSwitch
> 
>    v=0
>    o=- 3308986892 0 IN IP4 B-LEG_IP
>    s=Media Server
>    c=IN IP4 B-LEG_IP
>    t=0 0
>    m=audio 51246 RTP/AVP 0 101
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:20
>    ------------------------------------------------------------------------
> 
> Re-Invite from FS to A-Leg
>   ------------------------------------------------------------------------
>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>    Max-Forwards: 97
>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>    CSeq: 33480811 INVITE
>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>    User-Agent: FreeSwitch
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>    Require: timer
>    Supported: timer, precondition, path, replaces
>    Session-Expires: 64800;refresher=uas
>    Min-SE: 64800
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 235
>    X-FS-Support: update_display,send_info
> 
>    v=0
>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>    s=Media Server
>    c=IN IP4 B-LEG_IP
>    t=0 0
>    m=audio 51246 RTP/AVP 0 101
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:20
>    ------------------------------------------------------------------------
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mgg at giagnocavo.net  Sat Sep 15 02:48:37 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 14 Sep 2012 22:48:37 +0000
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAKDfjgcMA7e3Z22=3wcLWVyUz-Ygb3oejmRwGqfc=w_n1wiKJQ@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>
	<CAKDfjgcMA7e3Z22=3wcLWVyUz-Ygb3oejmRwGqfc=w_n1wiKJQ@mail.gmail.com>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FDF1CB@BLUPRD0711MB413.namprd07.prod.outlook.com>

I found someone with more info on scheduling problems, especially with sleep/wait, of which FS does a lot. There's a parameter called sched_migration_cost which might be worth investigating.

http://pastebin.freeswitch.com/19903

On a benchmark for sleeps, he says RHEL6 is taking up to 45% more time. With the huge number of threads and sleeps FS does, maybe those little differences add up:
http://unix.stackexchange.com/questions/37391/high-cpu-usage-with-cfs

Separately, to everyone that hasn't rebooted their server in the last few months, you may still have the leap-second set which will cause lots of CPU spinning due to a kernel bug. Running date -s "`date`" should fix that. I am pretty sure that freed up at least one core on one of our boxes that was still spinning since Jul 1.

We've now upgraded some systems to CentOS 6.3 (with all post release updates) and left others at 6.2. Over the next week we'll see if they seem to be much different. At the moment, the general CPU usage seems to be similar; we'll have to pound some dialer on it and see what happens.

-Michael


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: Monday, September 03, 2012 12:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

Anymore info on specific traffic patterns that cause this?  I've been beating up FS on 6.3 for several days now and haven't seen anything unusual yet...

On Thu, Aug 30, 2012 at 11:45 PM, Ken Rice <krice at freeswitch.org> wrote:
> 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...
>
>

--
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




From mgg at giagnocavo.net  Sat Sep 15 02:54:43 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 14 Sep 2012 22:54:43 +0000
Subject: [Freeswitch-users] Problem with originated calls hanging
	in	CS_SOFT_EXEC state
In-Reply-To: <20120904211433.ad092146@mail.tritonwest.net>
References: <6D0900A6-2881-47EF-8FEB-CC7354EA786E@vocalspace.com>
	<20120904211433.ad092146@mail.tritonwest.net>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FDF1F1@BLUPRD0711MB413.namprd07.prod.outlook.com>

If I can get some clear documentation on how the threading model is supposed to work, what's responsible for what, how a module can be re-entrant, etc. etc. I'll be happy to make nice clear managed code to make it easy.

Unfortunately, last time I got into it, it seemed to be rather complicated and not perfectly understood. And control modules were supposed to be of the "set a command and get lost" variety, and not really "own" the thread/call. There were also futher complications depending on if the call was bridged, and if it was bridged with media or not, and so on.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave R. Kompel
Sent: Tuesday, September 04, 2012 3:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state

Yes, they should be deprecated. Mod_managed has changed a lot over the years, and a lot of those things are still left over from the days when it was either single app domain, or there were no other for controling or collecting data form outbound legs.

One of the things i forgot to mention in my last reply, that is if you want to control an outbound legs completly form managed code, using the session object it is a lot easier to just do "Session.ExecuteString("originate channel/params &managed(yourclassname")". Then you can handle it just like at was an inbound call, and not even worry about clean-up.

The places you may want to use the hooks are for applications where you need to originate a call from API or other context, not actually have to control the call via managed code, but need to know about the outbound leg (execute_on_originate) where you can stash UUID, and other information about the leg, or get easy notification of call being answered/terminated (api_on_answer,api_hangup_hook). With both of these API hooks, set in the "originate" api call, you can also pass variable arguments that are expanded when they are executed {origination_nested_vars=true}. Does anyone know if that is documented on the wiki? I can't find it.

--Dave

________________________________
From: Phillip Boles [mailto:freeswitch-users at vocalspace.com]<mailto:[mailto:freeswitch-users at vocalspace.com]>
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]<mailto:[mailto:freeswitch-users at lists.freeswitch.org]>
Sent: Tue, 04 Sep 2012 13:35:47 -0700
Subject: Re: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state

Thanks Anthony! I found the ManageSession methods by looking at the code that is exposed via swig.

Using  "execute_on_*" hooks seems to be the proper procedure, it needs to be documented on the wiki. There is virtually no documentation to originate a call for mod managed on the wiki. I will correct that if I can get wiki access.

I will illustrate your solution using "execute_on_originate" as the preferred solution. There are also several "execute_on_*" hooks that would work.

Should ManagedSession.OriginateHandleHangup and .Originate() methods be deprecated to discourage their use in further releases? Or at least some comments in the code that point people to look at execute_on directives. Does Java have this same issue?

Anthony,Glad I found a bug by doing the wrong thing.... sorry for making work for ya!

I cannot say enough about how responsive Anthony and the community has been about this issue. THANKS!

On Sep 4, 2012, at 3:10 PM, Anthony Minessale wrote:


Listen to Dave.  I fixed the lock leak, it was down deep into code I
am not sure is ever used.   You want to be careful with what you do
especially since you are on linux which means you must be using Mono
which I am also not sure is used a lot.



On Tue, Sep 4, 2012 at 2:08 PM, Dave R. Kompel <drk at drkngs.net<mailto:drk at drkngs.net>> wrote:

This may be a bug, but ManagedSession really is not the right way to do this
from API context. Because of the APP_DOMAIN issue its much easier to just do
Api.ExecuteString("originate ... "). If you need to get to the "managed
session of the leg you have two options:

For getting to it before the originate set the variable execute_on_originate
to call a managed AppPlugin, or:

For getting results at the end of the call in API code, set the variable
API_HANGUP_HOOK.

Both of these methoods are much easier to do from API context in
mod_managed, and you won't have to worry about crossing app domain
boundries, and you won't have do do any cleanup on the leg.

________________________________
From: Anthony Minessale [mailto:anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>]
Sent: Tue, 04 Sep 2012 11:37:54 -0700

Subject: Re: [Freeswitch-users] Problem with originated calls hanging in
CS_SOFT_EXEC state

now that you have a jira do not continue this thread it doubles the
work effort, see comments there

On Tue, Sep 4, 2012 at 1:00 PM, Phillip Boles
<freeswitch-users at vocalspace.com<mailto:freeswitch-users at vocalspace.com>> wrote:
I have tried to get the Current HEAD to run mod_managed with this to the
top of switch_core.h
#define SWITCH_DEBUG_RWLOCKS 1

I am getting this error now.

2012-09-04 12:57:38.318859 [CRIT] switch_loadable_module.c:1310 Error
Loading module /usr/local/freeswitch/mod/mod_managed.so
**/usr/local/freeswitch/mod/mod_managed.so: undefined symbol:
switch_core_session_read_lock**

Thoughts?

Thanks!
On Aug 31, 2012, at 5:20 PM, Anthony Minessale wrote:

add this to the top of switch_core.h
#define SWITCH_DEBUG_RWLOCKS 1

rebuild and get a full log of the call
look for sign of unhandled rwlock

and put this on jira why I am i helping you over ml .... >=0


On Fri, Aug 31, 2012 at 4:16 PM, Phillip Boles
<freeswitch-users at vocalspace.com<mailto:freeswitch-users at vocalspace.com>> wrote:
Sorry Yes using the latest.

Using commit a8ce9ac29f3ba000bf42ab2286be04cc7bf9f509
Author: Anthony Minessale <anthm at freeswitch.org<mailto:anthm at freeswitch.org>>
Date: Thu Aug 30 17:17:15 2012 -0500

Changes made switch_cpp.cpp starting at Line 1000
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Calling
CoreSession::destroy\n");
if (session) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "We still have
valid session\n");

if (!channel) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
undefined! Trying to get it!\n");

channel = switch_core_session_get_channel(session);
}

if (channel) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
"%s destroy/unlink session from object\n",
switch_channel_get_name(channel));
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
"%s destroy/unlink uuid\n", switch_channel_get_uuid(channel));
switch_channel_set_private(channel, "CoreSession", NULL);
if (switch_channel_up(channel) && switch_test_flag(this, S_HUP) &&
!switch_channel_test_flag(channel, CF_TRANSFER)) {
switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Channel is
undefined! We Failed to get it!\n");
}

[CRIT] switch_cpp.cpp:1000 Calling CoreSession::destroy
[CRIT] switch_cpp.cpp:1002 We still have valid session
[DEBUG] switch_cpp.cpp:1011 sofia/external/12146635351 destroy/unlink
session from object
[DEBUG] switch_cpp.cpp:1013 83451093-e04f-49c1-9f55-5dd966bc4ba9
destroy/unlink uuid
[DEBUG] switch_core_state_machine.c:92 sofia/external/XXXXXXXXXX
Standard REPORTING, cause: NORMAL_CLEARING
[DEBUG] switch_core_state_machine.c:703 (sofia/external/XXXXXXXXXX )
State REPORTING going to sleep
[DEBUG] switch_core_state_machine.c:423 (sofia/external/XXXXXXXXXX )
State Change CS_REPORTING -> CS_DESTROY
[DEBUG] switch_core_session.c:1210 Send signal sofia/external/XXXXXXXXXX
[BREAK]
[DEBUG] switch_core_session.c:1415 Session 2 (sofia/external/XXXXXXXXXX
) Locked, Waiting on external entities



channel remains hung

P

On Aug 31, 2012, at 12:26 PM, Anthony Minessale wrote:

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<mailto:freeswitch-users at vocalspace.com>> wrote:
var session = ManagedSession.OriginateHandleHangup(s, dialString, ts,
(y)
=> {
try {
Log.WriteLine(LogLevel.Info<http://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<mailto: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<mailto: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<mailto:anthony.minessale at gmail.com>]
To: FreeSWITCH Users Help
[mailto:freeswitch-users at lists.freeswitch.org<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<mailto: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<http://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 msc at freeswitch.org  Sat Sep 15 03:20:48 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Sep 2012 16:20:48 -0700
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
Message-ID: <CAKzWOxXZgrGVRjgyOhKLdnjB6gmM=vGOViTc65ZjqhhocDGheQ@mail.gmail.com>

Thanks for the follow up. I don't know the answer to either of your
questions so I'll defer to those with more experience than I.
-MC

On Fri, Sep 14, 2012 at 10:54 AM, Peter van Raamsdonk <peetzer at gmail.com>wrote:

> Hi Michael and group,
>
> I tried the mod_xml_curl after studying the wiki. It works like a charm
> and FS post to my java servlet easily (dialplan and user directory).
> I read on a blog there is option 'cacheable=true' to prevent a post every
> time a dial is made, do you know where to put this?
>
> I suspected that the reloadxml would trigger a post but it doesn't,
> dialplan is triggered from dial and user directory from startup of FS and
> dial I thought.
>
> FS is highly customizable and I need to find my way but it works like a
> charm.
>
> I wonder if there are people who had a successful mod_xml_java build on
> Windows. I made a own project and put references to the core but got stuck
> and left it because it took too much time.
>
> Thanks and have a nice weekend all!
>
> Peter
>
> On 14 sep 2012, at 19:01, Michael Collins wrote:
>
> Hi Peter!
>
> Welcome to FreeSWITCH! Glad to have you. These are great questions. FS has
> a db command that can be executed from fs_cli, however this isn't what you
> want/need for users. The user directory is where all that stuff is handled.
> Static XML is, of course, in conf/directory/* and you can add/remove there,
> but that doesn't sound like what you need. The best option for you right
> now is to get up to speed with mod_xml_curl. Using mod_xml_curl you can
> have FS poll a database each time there is a need, such as when there is an
> auth challenge taking place.
>
> We have several people who've done lots of mod_xml_curl stuff, so feel
> free to ask away, but please check the wiki first since there is a lot of
> information and examples there. You may also enjoy discussing this live in
> #freeswitch channel on irc.freenode.net.
>
> -MC
>
> On Fri, Sep 14, 2012 at 2:03 AM, Peter van Raamsdonk <peetzer at gmail.com>wrote:
>
>> Hi all,
>>
>> Just wondering if I can add/ delete users to the db from the CLI (I'm
>> using ESL java library)?
>>
>> See http://wiki.freeswitch.org/wiki/Function_db
>>
>> From the Dialplan this is possible, I want however to dynamically add new
>> users.
>>
>> <action application="db" data="insert/realm/key/value"/>
>>
>>
>> In the api (mod commands) nothing is mentioned about "db"
>>
>> I don't know if something like this could work with my example extension
>> 1999, default group named default...
>>
>> response = client.sendSyncApiCommand( "db", "insert/default/1999/1999");
>>
>> Another option is to use mod_xml_curl and feed the dialplan or user
>> directory to FreeSwitch back again.
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 msc at freeswitch.org  Sat Sep 15 03:42:37 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Sep 2012 16:42:37 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
Message-ID: <CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>

Mike, try comma separating the values instead of using two sets of {}:
add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");

-MC

On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> I have that set in the bridge command however does not seem to be working
> as expected.
>
>
> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>
>
>
> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> set {sip_require_timer=false} in your outbound calls or globally
>
>
> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> it seems if I get an re-invite from the B-Leg FS add's requires timer and
>> changes the session timer to a high value to the re-invite going to the
>> A-Leg come to find out Acme Packets at our ULC's do not like this and send
>> us back a 420 Bad Extension and the call is disconnected with Reason:
>> Q.850;cause=127;text="INTERWORKING"
>>
>> The ULC are stating we need to only have it in our supported and do not
>> pass them a require.
>>
>> Re-Invite from B-Leg to FS
>>
>>  ------------------------------------------------------------------------
>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>    Via: SIP/2.0/UDP
>> B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>    Content-Length: 217
>>    Content-Type: application/sdp
>>    CSeq: 33480808 INVITE
>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>    Max-Forwards: 92
>>    Session-Expires: 3600;refresher=uas
>>    Supported: timer
>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>    User-Agent: FreeSwitch
>>
>>    v=0
>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>    s=Media Server
>>    c=IN IP4 B-LEG_IP
>>    t=0 0
>>    m=audio 51246 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:20
>>
>>  ------------------------------------------------------------------------
>>
>> Re-Invite from FS to A-Leg
>>   ------------------------------------------------------------------------
>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>    Max-Forwards: 97
>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy
>> :5060;user=phone>;tag=c9Favaa53XFXB
>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER
>> :5060;user=phone>;tag=SDd626401-gK095bbb72
>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>    CSeq: 33480811 INVITE
>>    Contact: <
>> sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>    User-Agent: FreeSwitch
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>    Require: timer
>>    Supported: timer, precondition, path, replaces
>>    Session-Expires: 64800;refresher=uas
>>    Min-SE: 64800
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 235
>>    X-FS-Support: update_display,send_info
>>
>>    v=0
>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>    s=Media Server
>>    c=IN IP4 B-LEG_IP
>>    t=0 0
>>    m=audio 51246 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:20
>>
>>  ------------------------------------------------------------------------
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>


-- 
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  Sat Sep 15 04:16:32 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Sep 2012 19:16:32 -0500
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
Message-ID: <CAKbxfG_Bm8k8hG_H_Boey21y9kdARYe=OdH0zR3bfp-Je51LQg@mail.gmail.com>

If its b leg sending invite you need to set it on that leg, so in this case
set it on the so too or just globally set it in vars.xml and forget about
it and it will apply to all legs which should be ok.  This particular case
is a bot of a philosophical debate on proper action as we have devices
break on it going both ways hence the setting.
On Sep 14, 2012 6:44 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Mike, try comma separating the values instead of using two sets of {}:
>
> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>
> -MC
>
> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> I have that set in the bridge command however does not seem to be working
>> as expected.
>>
>>
>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>
>>
>>
>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> set {sip_require_timer=false} in your outbound calls or globally
>>
>>
>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>>
>>> it seems if I get an re-invite from the B-Leg FS add's requires timer
>>> and changes the session timer to a high value to the re-invite going to the
>>> A-Leg come to find out Acme Packets at our ULC's do not like this and send
>>> us back a 420 Bad Extension and the call is disconnected with Reason:
>>> Q.850;cause=127;text="INTERWORKING"
>>>
>>> The ULC are stating we need to only have it in our supported and do not
>>> pass them a require.
>>>
>>> Re-Invite from B-Leg to FS
>>>
>>>  ------------------------------------------------------------------------
>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>    Via: SIP/2.0/UDP
>>> B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>    Content-Length: 217
>>>    Content-Type: application/sdp
>>>    CSeq: 33480808 INVITE
>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>    Max-Forwards: 92
>>>    Session-Expires: 3600;refresher=uas
>>>    Supported: timer
>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>    User-Agent: FreeSwitch
>>>
>>>    v=0
>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>
>>>  ------------------------------------------------------------------------
>>>
>>> Re-Invite from FS to A-Leg
>>>
>>> ------------------------------------------------------------------------
>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>    Max-Forwards: 97
>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy
>>> :5060;user=phone>;tag=c9Favaa53XFXB
>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER
>>> :5060;user=phone>;tag=SDd626401-gK095bbb72
>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>    CSeq: 33480811 INVITE
>>>    Contact: <
>>> sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>    User-Agent: FreeSwitch
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> NOTIFY
>>>    Require: timer
>>>    Supported: timer, precondition, path, replaces
>>>    Session-Expires: 64800;refresher=uas
>>>    Min-SE: 64800
>>>    Content-Type: application/sdp
>>>    Content-Disposition: session
>>>    Content-Length: 235
>>>    X-FS-Support: update_display,send_info
>>>
>>>    v=0
>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>
>>>  ------------------------------------------------------------------------
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>
>
> --
> 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 mike.burlingame at me.com  Sat Sep 15 04:18:15 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 17:18:15 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
Message-ID: <58797397-E56A-4B70-AF67-3203EBC0109E@me.com>

Thanks that seem to help the high session timer value however the require: timer is still present 

Invite from B-Leg --> FreeSwitch
recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
   ------------------------------------------------------------------------
   INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
   Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
   Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
   Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
   Max-Forwards: 68
   Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
   From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
   To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
   CSeq: 219015274 INVITE
   Expires: 180
   Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
   Organization: MetaSwitch
   Supported: 100rel, resource-priority
   Content-Length: 193
   Content-Type: application/sdp
   
   v=0
   o=- 3341834898 3341834898 IN IP4 Carrier_IP
   s=-
   c=IN IP4 Carrier_IP
   t=0 0
   m=audio 29864 RTP/AVP 18 0 101
   a=rtpmap:101 telephone-event/8000
   a=ptime:20
   a=silenceSupp:off - - - -



Invite from FreeSwitch --> A-Leg
send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
   ------------------------------------------------------------------------
   INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
   Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
   Route: <sip:ULC;lr=on>
   Max-Forwards: 47
   From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
   To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
   Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
   CSeq: 33497250 INVITE
   Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
   User-Agent: FreeSwitch
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
   Require: timer
   Supported: timer, precondition, path, replaces
   Session-Expires: 3600;refresher=uas
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 193
   P-hint: outbound
   X-FS-Support: update_display,send_info
   
   v=0
   o=- 3341834898 3341834898 IN IP4 Carrier_IP
   s=-
   c=IN IP4 Carrier_IP
   t=0 0
   m=audio 29864 RTP/AVP 18 0 101
   a=rtpmap:101 telephone-event/8000
   a=ptime:20
   a=silenceSupp:off - - - -


On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:

> Mike, try comma separating the values instead of using two sets of {}:
> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
> 
> -MC
> 
> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> I have that set in the bridge command however does not seem to be working as expected.
> 
> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
> 
> 
> 
> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> set {sip_require_timer=false} in your outbound calls or globally 
>> 
>> 
>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
>> 
>> The ULC are stating we need to only have it in our supported and do not pass them a require.
>> 
>> Re-Invite from B-Leg to FS
>>    ------------------------------------------------------------------------
>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>    Content-Length: 217
>>    Content-Type: application/sdp
>>    CSeq: 33480808 INVITE
>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>    Max-Forwards: 92
>>    Session-Expires: 3600;refresher=uas
>>    Supported: timer
>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>    User-Agent: FreeSwitch
>> 
>>    v=0
>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>    s=Media Server
>>    c=IN IP4 B-LEG_IP
>>    t=0 0
>>    m=audio 51246 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:20
>>    ------------------------------------------------------------------------
>> 
>> Re-Invite from FS to A-Leg
>>   ------------------------------------------------------------------------
>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>    Max-Forwards: 97
>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>    CSeq: 33480811 INVITE
>>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>    User-Agent: FreeSwitch
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>    Require: timer
>>    Supported: timer, precondition, path, replaces
>>    Session-Expires: 64800;refresher=uas
>>    Min-SE: 64800
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 235
>>    X-FS-Support: update_display,send_info
>> 
>>    v=0
>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>    s=Media Server
>>    c=IN IP4 B-LEG_IP
>>    t=0 0
>>    m=audio 51246 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:20
>>    ------------------------------------------------------------------------
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> 
> 
> 
> 
> -- 
> 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 mike.burlingame at me.com  Sat Sep 15 04:28:06 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 17:28:06 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKbxfG_Bm8k8hG_H_Boey21y9kdARYe=OdH0zR3bfp-Je51LQg@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<CAKbxfG_Bm8k8hG_H_Boey21y9kdARYe=OdH0zR3bfp-Je51LQg@mail.gmail.com>
Message-ID: <8AED7F70-A57A-4659-B19E-F947DD1BC7BD@me.com>

gonna set it in vars.xml and call it a day and see if it works

On Sep 14, 2012, at 5:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> If its b leg sending invite you need to set it on that leg, so in this case set it on the so too or just globally set it in vars.xml and forget about it and it will apply to all legs which should be ok.  This particular case is a bot of a philosophical debate on proper action as we have devices break on it going both ways hence the setting.
> 
> On Sep 14, 2012 6:44 PM, "Michael Collins" <msc at freeswitch.org> wrote:
> Mike, try comma separating the values instead of using two sets of {}:
> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
> 
> -MC
> 
> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> I have that set in the bridge command however does not seem to be working as expected.
> 
> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
> 
> 
> 
> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> set {sip_require_timer=false} in your outbound calls or globally 
>> 
>> 
>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
>> 
>> The ULC are stating we need to only have it in our supported and do not pass them a require.
>> 
>> Re-Invite from B-Leg to FS
>>    ------------------------------------------------------------------------
>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>    Content-Length: 217
>>    Content-Type: application/sdp
>>    CSeq: 33480808 INVITE
>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>    Max-Forwards: 92
>>    Session-Expires: 3600;refresher=uas
>>    Supported: timer
>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>    User-Agent: FreeSwitch
>> 
>>    v=0
>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>    s=Media Server
>>    c=IN IP4 B-LEG_IP
>>    t=0 0
>>    m=audio 51246 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:20
>>    ------------------------------------------------------------------------
>> 
>> Re-Invite from FS to A-Leg
>>   ------------------------------------------------------------------------
>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>    Max-Forwards: 97
>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>    CSeq: 33480811 INVITE
>>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>    User-Agent: FreeSwitch
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>    Require: timer
>>    Supported: timer, precondition, path, replaces
>>    Session-Expires: 64800;refresher=uas
>>    Min-SE: 64800
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 235
>>    X-FS-Support: update_display,send_info
>> 
>>    v=0
>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>    s=Media Server
>>    c=IN IP4 B-LEG_IP
>>    t=0 0
>>    m=audio 51246 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:20
>>    ------------------------------------------------------------------------
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> 
> 
> 
> 
> -- 
> 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 anthony.minessale at gmail.com  Sat Sep 15 04:29:14 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Sep 2012 19:29:14 -0500
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
Message-ID: <CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>

It's Wong to not accept it. IIRC...there is a better case for why it should
be there than for it not being there.  It's a rather dumb thing for some of
these commercial switches to even care about....I don't want to set a
precedence here that I will perpetually change our sip stack and chase
after every interop case... Make sure the param is set, I believe it
disables it completely going against the rfc but I can't check till
Monday.  This is part of Sofia not FS itself.
On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:

> Thanks that seem to help the high session timer value however the require:
> timer is still present
>
> Invite from B-Leg --> FreeSwitch
> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>    ------------------------------------------------------------------------
>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>    Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>    Via: SIP/2.0/UDP
> Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>    Allow-Events: message-summary, refer, dialog, line-seize, presence,
> call-info, as-feature-event
>    Max-Forwards: 68
>    Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>    From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>    CSeq: 219015274 INVITE
>    Expires: 180
>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>    Organization: MetaSwitch
>    Supported: 100rel, resource-priority
>    Content-Length: 193
>    Content-Type: application/sdp
>
>    v=0
>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>    s=-
>    c=IN IP4 Carrier_IP
>    t=0 0
>    m=audio 29864 RTP/AVP 18 0 101
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=silenceSupp:off - - - -
>
>
>
> Invite from FreeSwitch --> A-Leg
> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>    ------------------------------------------------------------------------
>    INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>    Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>    Route: <sip:ULC;lr=on>
>    Max-Forwards: 47
>    From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>    CSeq: 33497250 INVITE
>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>    User-Agent: FreeSwitch
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>    Require: timer
>    Supported: timer, precondition, path, replaces
>    Session-Expires: 3600;refresher=uas
>    Min-SE: 120
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 193
>    P-hint: outbound
>    X-FS-Support: update_display,send_info
>
>    v=0
>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>    s=-
>    c=IN IP4 Carrier_IP
>    t=0 0
>    m=audio 29864 RTP/AVP 18 0 101
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=silenceSupp:off - - - -
>
>
> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> Mike, try comma separating the values instead of using two sets of {}:
>
> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>
> -MC
>
> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> I have that set in the bridge command however does not seem to be working
>> as expected.
>>
>>
>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>
>>
>>
>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> set {sip_require_timer=false} in your outbound calls or globally
>>
>>
>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>>
>>> it seems if I get an re-invite from the B-Leg FS add's requires timer
>>> and changes the session timer to a high value to the re-invite going to the
>>> A-Leg come to find out Acme Packets at our ULC's do not like this and send
>>> us back a 420 Bad Extension and the call is disconnected with Reason:
>>> Q.850;cause=127;text="INTERWORKING"
>>>
>>> The ULC are stating we need to only have it in our supported and do not
>>> pass them a require.
>>>
>>> Re-Invite from B-Leg to FS
>>>
>>>  ------------------------------------------------------------------------
>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>    Via: SIP/2.0/UDP
>>> B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>    Content-Length: 217
>>>    Content-Type: application/sdp
>>>    CSeq: 33480808 INVITE
>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>    Max-Forwards: 92
>>>    Session-Expires: 3600;refresher=uas
>>>    Supported: timer
>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>    User-Agent: FreeSwitch
>>>
>>>    v=0
>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>
>>>  ------------------------------------------------------------------------
>>>
>>> Re-Invite from FS to A-Leg
>>>
>>> ------------------------------------------------------------------------
>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>    Max-Forwards: 97
>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy
>>> :5060;user=phone>;tag=c9Favaa53XFXB
>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER
>>> :5060;user=phone>;tag=SDd626401-gK095bbb72
>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>    CSeq: 33480811 INVITE
>>>    Contact: <
>>> sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>    User-Agent: FreeSwitch
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> NOTIFY
>>>    Require: timer
>>>    Supported: timer, precondition, path, replaces
>>>    Session-Expires: 64800;refresher=uas
>>>    Min-SE: 64800
>>>    Content-Type: application/sdp
>>>    Content-Disposition: session
>>>    Content-Length: 235
>>>    X-FS-Support: update_display,send_info
>>>
>>>    v=0
>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>
>>>  ------------------------------------------------------------------------
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <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 mike.burlingame at me.com  Sat Sep 15 04:37:28 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 17:37:28 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
	<CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
Message-ID: <1EC0A47E-496E-4F2C-934A-4626AFB86F7E@me.com>

I agree if it's in the supported line why can it not be required and I am fighting with PaeTec and Global Crossing right now on this issue - They want me to just "Disable" it but I do not feel that is the proper thing to do however to resolve the 420's I might have to give in till I can get them to accept it and not fail on it

Both carriers use Acme Packet as the SBC and I have a feeling this is where the hangup is coming in at 


On Sep 14, 2012, at 5:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> It's Wong to not accept it. IIRC...there is a better case for why it should be there than for it not being there.  It's a rather dumb thing for some of these commercial switches to even care about....I don't want to set a precedence here that I will perpetually change our sip stack and chase after every interop case... Make sure the param is set, I believe it disables it completely going against the rfc but I can't check till Monday.  This is part of Sofia not FS itself.
> 
> On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
> Thanks that seem to help the high session timer value however the require: timer is still present 
> 
> Invite from B-Leg --> FreeSwitch
> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>    ------------------------------------------------------------------------
>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>    Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>    Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>    Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
>    Max-Forwards: 68
>    Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>    From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>    CSeq: 219015274 INVITE
>    Expires: 180
>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>    Organization: MetaSwitch
>    Supported: 100rel, resource-priority
>    Content-Length: 193
>    Content-Type: application/sdp
>    
>    v=0
>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>    s=-
>    c=IN IP4 Carrier_IP
>    t=0 0
>    m=audio 29864 RTP/AVP 18 0 101
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=silenceSupp:off - - - -
> 
> 
> 
> Invite from FreeSwitch --> A-Leg
> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>    ------------------------------------------------------------------------
>    INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>    Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>    Route: <sip:ULC;lr=on>
>    Max-Forwards: 47
>    From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>    CSeq: 33497250 INVITE
>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>    User-Agent: FreeSwitch
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>    Require: timer
>    Supported: timer, precondition, path, replaces
>    Session-Expires: 3600;refresher=uas
>    Min-SE: 120
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 193
>    P-hint: outbound
>    X-FS-Support: update_display,send_info
>    
>    v=0
>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>    s=-
>    c=IN IP4 Carrier_IP
>    t=0 0
>    m=audio 29864 RTP/AVP 18 0 101
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=silenceSupp:off - - - -
> 
> 
> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> Mike, try comma separating the values instead of using two sets of {}:
>> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>> 
>> -MC
>> 
>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> I have that set in the bridge command however does not seem to be working as expected.
>> 
>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>> 
>> 
>> 
>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> set {sip_require_timer=false} in your outbound calls or globally 
>>> 
>>> 
>>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
>>> 
>>> The ULC are stating we need to only have it in our supported and do not pass them a require.
>>> 
>>> Re-Invite from B-Leg to FS
>>>    ------------------------------------------------------------------------
>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>    Content-Length: 217
>>>    Content-Type: application/sdp
>>>    CSeq: 33480808 INVITE
>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>    Max-Forwards: 92
>>>    Session-Expires: 3600;refresher=uas
>>>    Supported: timer
>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>    User-Agent: FreeSwitch
>>> 
>>>    v=0
>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>    ------------------------------------------------------------------------
>>> 
>>> Re-Invite from FS to A-Leg
>>>   ------------------------------------------------------------------------
>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>    Max-Forwards: 97
>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>    CSeq: 33480811 INVITE
>>>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>    User-Agent: FreeSwitch
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>>    Require: timer
>>>    Supported: timer, precondition, path, replaces
>>>    Session-Expires: 64800;refresher=uas
>>>    Min-SE: 64800
>>>    Content-Type: application/sdp
>>>    Content-Disposition: session
>>>    Content-Length: 235
>>>    X-FS-Support: update_display,send_info
>>> 
>>>    v=0
>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>    ------------------------------------------------------------------------
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
>> 
>> 
>> 
>> -- 
>> 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

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From mike.burlingame at me.com  Sat Sep 15 04:40:14 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 17:40:14 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <1EC0A47E-496E-4F2C-934A-4626AFB86F7E@me.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
	<CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
	<1EC0A47E-496E-4F2C-934A-4626AFB86F7E@me.com>
Message-ID: <3D18A14B-6820-437A-B878-6445CC3FB8BB@me.com>

However and oddly enough Level(3)'s Sonus platform deals with this just fine


On Sep 14, 2012, at 5:37 PM, Mike Burlingame <mike.burlingame at me.com> wrote:

> I agree if it's in the supported line why can it not be required and I am fighting with PaeTec and Global Crossing right now on this issue - They want me to just "Disable" it but I do not feel that is the proper thing to do however to resolve the 420's I might have to give in till I can get them to accept it and not fail on it
> 
> Both carriers use Acme Packet as the SBC and I have a feeling this is where the hangup is coming in at 
> 
> 
> On Sep 14, 2012, at 5:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> It's Wong to not accept it. IIRC...there is a better case for why it should be there than for it not being there.  It's a rather dumb thing for some of these commercial switches to even care about....I don't want to set a precedence here that I will perpetually change our sip stack and chase after every interop case... Make sure the param is set, I believe it disables it completely going against the rfc but I can't check till Monday.  This is part of Sofia not FS itself.
>> 
>> On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
>> Thanks that seem to help the high session timer value however the require: timer is still present 
>> 
>> Invite from B-Leg --> FreeSwitch
>> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>>    ------------------------------------------------------------------------
>>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>>    Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>>    Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>>    Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
>>    Max-Forwards: 68
>>    Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>>    From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>>    CSeq: 219015274 INVITE
>>    Expires: 180
>>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>>    Organization: MetaSwitch
>>    Supported: 100rel, resource-priority
>>    Content-Length: 193
>>    Content-Type: application/sdp
>>    
>>    v=0
>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>    s=-
>>    c=IN IP4 Carrier_IP
>>    t=0 0
>>    m=audio 29864 RTP/AVP 18 0 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=ptime:20
>>    a=silenceSupp:off - - - -
>> 
>> 
>> 
>> Invite from FreeSwitch --> A-Leg
>> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>>    ------------------------------------------------------------------------
>>    INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>>    Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>>    Route: <sip:ULC;lr=on>
>>    Max-Forwards: 47
>>    From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>>    CSeq: 33497250 INVITE
>>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>>    User-Agent: FreeSwitch
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>    Require: timer
>>    Supported: timer, precondition, path, replaces
>>    Session-Expires: 3600;refresher=uas
>>    Min-SE: 120
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 193
>>    P-hint: outbound
>>    X-FS-Support: update_display,send_info
>>    
>>    v=0
>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>    s=-
>>    c=IN IP4 Carrier_IP
>>    t=0 0
>>    m=audio 29864 RTP/AVP 18 0 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=ptime:20
>>    a=silenceSupp:off - - - -
>> 
>> 
>> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> Mike, try comma separating the values instead of using two sets of {}:
>>> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>> 
>>> -MC
>>> 
>>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> I have that set in the bridge command however does not seem to be working as expected.
>>> 
>>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>> 
>>> 
>>> 
>>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> 
>>>> set {sip_require_timer=false} in your outbound calls or globally 
>>>> 
>>>> 
>>>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
>>>> 
>>>> The ULC are stating we need to only have it in our supported and do not pass them a require.
>>>> 
>>>> Re-Invite from B-Leg to FS
>>>>    ------------------------------------------------------------------------
>>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>>    Content-Length: 217
>>>>    Content-Type: application/sdp
>>>>    CSeq: 33480808 INVITE
>>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>>    Max-Forwards: 92
>>>>    Session-Expires: 3600;refresher=uas
>>>>    Supported: timer
>>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>>    User-Agent: FreeSwitch
>>>> 
>>>>    v=0
>>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>>    s=Media Server
>>>>    c=IN IP4 B-LEG_IP
>>>>    t=0 0
>>>>    m=audio 51246 RTP/AVP 0 101
>>>>    a=rtpmap:0 PCMU/8000
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-15
>>>>    a=ptime:20
>>>>    ------------------------------------------------------------------------
>>>> 
>>>> Re-Invite from FS to A-Leg
>>>>   ------------------------------------------------------------------------
>>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>>    Max-Forwards: 97
>>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>>    CSeq: 33480811 INVITE
>>>>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>>    User-Agent: FreeSwitch
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>>>    Require: timer
>>>>    Supported: timer, precondition, path, replaces
>>>>    Session-Expires: 64800;refresher=uas
>>>>    Min-SE: 64800
>>>>    Content-Type: application/sdp
>>>>    Content-Disposition: session
>>>>    Content-Length: 235
>>>>    X-FS-Support: update_display,send_info
>>>> 
>>>>    v=0
>>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>>    s=Media Server
>>>>    c=IN IP4 B-LEG_IP
>>>>    t=0 0
>>>>    m=audio 51246 RTP/AVP 0 101
>>>>    a=rtpmap:0 PCMU/8000
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-15
>>>>    a=ptime:20
>>>>    ------------------------------------------------------------------------
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>> 
>>> 
>>> 
>>> 
>>> -- 
>>> 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 15 04:47:41 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 14 Sep 2012 17:47:41 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
	<CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
Message-ID: <0D4E3E2F-C975-4826-B50F-6CE09D3E520B@me.com>

This worked when set in the vars.xml if you can please let me know on monday what you find that would be great for the time being I am going to keep pushing GC and PaeTec to fix it's config and or work with it's vendors.

On Sep 14, 2012, at 5:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> It's Wong to not accept it. IIRC...there is a better case for why it should be there than for it not being there.  It's a rather dumb thing for some of these commercial switches to even care about....I don't want to set a precedence here that I will perpetually change our sip stack and chase after every interop case... Make sure the param is set, I believe it disables it completely going against the rfc but I can't check till Monday.  This is part of Sofia not FS itself.
> 
> On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
> Thanks that seem to help the high session timer value however the require: timer is still present 
> 
> Invite from B-Leg --> FreeSwitch
> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>    ------------------------------------------------------------------------
>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>    Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>    Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>    Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
>    Max-Forwards: 68
>    Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>    From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>    CSeq: 219015274 INVITE
>    Expires: 180
>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>    Organization: MetaSwitch
>    Supported: 100rel, resource-priority
>    Content-Length: 193
>    Content-Type: application/sdp
>    
>    v=0
>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>    s=-
>    c=IN IP4 Carrier_IP
>    t=0 0
>    m=audio 29864 RTP/AVP 18 0 101
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=silenceSupp:off - - - -
> 
> 
> 
> Invite from FreeSwitch --> A-Leg
> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>    ------------------------------------------------------------------------
>    INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>    Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>    Route: <sip:ULC;lr=on>
>    Max-Forwards: 47
>    From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>    CSeq: 33497250 INVITE
>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>    User-Agent: FreeSwitch
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>    Require: timer
>    Supported: timer, precondition, path, replaces
>    Session-Expires: 3600;refresher=uas
>    Min-SE: 120
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 193
>    P-hint: outbound
>    X-FS-Support: update_display,send_info
>    
>    v=0
>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>    s=-
>    c=IN IP4 Carrier_IP
>    t=0 0
>    m=audio 29864 RTP/AVP 18 0 101
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=silenceSupp:off - - - -
> 
> 
> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> Mike, try comma separating the values instead of using two sets of {}:
>> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>> 
>> -MC
>> 
>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> I have that set in the bridge command however does not seem to be working as expected.
>> 
>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>> 
>> 
>> 
>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> set {sip_require_timer=false} in your outbound calls or globally 
>>> 
>>> 
>>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
>>> 
>>> The ULC are stating we need to only have it in our supported and do not pass them a require.
>>> 
>>> Re-Invite from B-Leg to FS
>>>    ------------------------------------------------------------------------
>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>    Content-Length: 217
>>>    Content-Type: application/sdp
>>>    CSeq: 33480808 INVITE
>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>    Max-Forwards: 92
>>>    Session-Expires: 3600;refresher=uas
>>>    Supported: timer
>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>    User-Agent: FreeSwitch
>>> 
>>>    v=0
>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>    ------------------------------------------------------------------------
>>> 
>>> Re-Invite from FS to A-Leg
>>>   ------------------------------------------------------------------------
>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>    Max-Forwards: 97
>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>    CSeq: 33480811 INVITE
>>>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>    User-Agent: FreeSwitch
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>>    Require: timer
>>>    Supported: timer, precondition, path, replaces
>>>    Session-Expires: 64800;refresher=uas
>>>    Min-SE: 64800
>>>    Content-Type: application/sdp
>>>    Content-Disposition: session
>>>    Content-Length: 235
>>>    X-FS-Support: update_display,send_info
>>> 
>>>    v=0
>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>    s=Media Server
>>>    c=IN IP4 B-LEG_IP
>>>    t=0 0
>>>    m=audio 51246 RTP/AVP 0 101
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=ptime:20
>>>    ------------------------------------------------------------------------
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
>> 
>> 
>> 
>> -- 
>> 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

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From anton.jugatsu at gmail.com  Sat Sep 15 10:28:11 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sat, 15 Sep 2012 10:28:11 +0400
Subject: [Freeswitch-users] 481 Call Leg/Transaction Does Not Exist
In-Reply-To: <99D9E8DC-8621-47E4-8F25-44F7B33001B2@kavun.ch>
References: <5D5FD0B1-34C0-4C06-AD20-735E3ACE8803@kavun.ch>
	<99D9E8DC-8621-47E4-8F25-44F7B33001B2@kavun.ch>
Message-ID: <CANBhyPAD7kuZ8_e8KoWSxvGpeaEh=oNkqWn89MNVGMqYMFzb2Q@mail.gmail.com>

Emrah, can you please paste the successful dump of call when you transfer
from Polycom. I observe strange bahaviour when FS sends BYE to Yealink to
shutdown the dialog and Yealink also kills the dialog with it's BYE.

2012/9/15 Emrah <lists at kavun.ch>

> Hi all,
>
> I have tried to play with the refer extension and now have my calls land
> in the public context when they are transferred?
> This is the second extension in my dialplan:
> <extension name="refer">
>  <condition field="${sip_refer_to}">
>         <expression><![CDATA[<sip:${destination_number}@
> ${domain_name}>]]></expression>
>  </condition>
>  <condition field="${sip_refer_to}">
>         <expression><![CDATA[<sip:(.*)@(.*)>]]></expression>
>         <action application="set" data="refer_user=$1"/>
>         <action application="set" data="refer_domain=$2"/>
>         <action application="info"/>
>         <action application="bridge"
> data="sofia/${use_profile}/${refer_user}@${domain_name}"/>
>  </condition>
>     </extension>
>
> Any help would be greatly appreciated.
>
> All the best,
> Emrah
> On Sep 13, 2012, at 3:09 PM, Emrah <lists at kavun.ch> wrote:
>
> > Hi all,
> >
> > I seem to be encountering the strangest problems.
> > I have been doing some more testing with my transfers and here is a
> recurring issue.
> >
> > I have 2 phones behind the same router. 1 Polycom VVX 1500 and 1 Yealink
> SIP-T38g
> > They are both registering on the same FS server which is on a remote
> public IP.
> > The Polycom has extension 20 and the yealink uses 21.
> > When I call the Polycom from the Yealink and perform a blind transfer
> from the Yealink, it fails with: 481 Call Leg/Transaction Does Not Exist.
> > If I transfer from the Polycom though, it seems to work.
> >
> > In resume:
> > I cannot transfer from A leg. B leg doesn't seem to know its way around
> and looses it. To demonstrate, I transfer B leg from A leg to extension
> 5000.
> > http://pastebin.freeswitch.org/19893
> >
> > I tried Googleing the error code and didn't find anything particularly
> related to my issue.
> >
> > Any hint would be greatly appreciated. To make sure, I downgraded to the
> stable version.
> >
> > Thanks a bunch, as always.
> > 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 anton.jugatsu at gmail.com  Sat Sep 15 11:11:52 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sat, 15 Sep 2012 11:11:52 +0400
Subject: [Freeswitch-users] 481 Call Leg/Transaction Does Not Exist
In-Reply-To: <CANBhyPAD7kuZ8_e8KoWSxvGpeaEh=oNkqWn89MNVGMqYMFzb2Q@mail.gmail.com>
References: <5D5FD0B1-34C0-4C06-AD20-735E3ACE8803@kavun.ch>
	<99D9E8DC-8621-47E4-8F25-44F7B33001B2@kavun.ch>
	<CANBhyPAD7kuZ8_e8KoWSxvGpeaEh=oNkqWn89MNVGMqYMFzb2Q@mail.gmail.com>
Message-ID: <CANBhyPBjH_--apWHb5aGAkNy61Mhnmx6AK-i5H1Zesx352pUTQ@mail.gmail.com>

So the question is: why the hack FS sends BYE to Yealink to tear down the
session when this must do the UA which is performing the call transfer with
REFER method. Obviously, Yealink responds with 481 to this kind of BYE,
becauce it has already killed the dialog.

2012/9/15 Anton Kvashenkin <anton.jugatsu at gmail.com>

> Emrah, can you please paste the successful dump of call when you transfer
> from Polycom. I observe strange bahaviour when FS sends BYE to Yealink to
> shutdown the dialog and Yealink also kills the dialog with it's BYE.
>
>
> 2012/9/15 Emrah <lists at kavun.ch>
>
>> Hi all,
>>
>> I have tried to play with the refer extension and now have my calls land
>> in the public context when they are transferred?
>> This is the second extension in my dialplan:
>> <extension name="refer">
>>  <condition field="${sip_refer_to}">
>>         <expression><![CDATA[<sip:${destination_number}@
>> ${domain_name}>]]></expression>
>>  </condition>
>>  <condition field="${sip_refer_to}">
>>         <expression><![CDATA[<sip:(.*)@(.*)>]]></expression>
>>         <action application="set" data="refer_user=$1"/>
>>         <action application="set" data="refer_domain=$2"/>
>>         <action application="info"/>
>>         <action application="bridge"
>> data="sofia/${use_profile}/${refer_user}@${domain_name}"/>
>>  </condition>
>>     </extension>
>>
>> Any help would be greatly appreciated.
>>
>> All the best,
>> Emrah
>> On Sep 13, 2012, at 3:09 PM, Emrah <lists at kavun.ch> wrote:
>>
>> > Hi all,
>> >
>> > I seem to be encountering the strangest problems.
>> > I have been doing some more testing with my transfers and here is a
>> recurring issue.
>> >
>> > I have 2 phones behind the same router. 1 Polycom VVX 1500 and 1
>> Yealink SIP-T38g
>> > They are both registering on the same FS server which is on a remote
>> public IP.
>> > The Polycom has extension 20 and the yealink uses 21.
>> > When I call the Polycom from the Yealink and perform a blind transfer
>> from the Yealink, it fails with: 481 Call Leg/Transaction Does Not Exist.
>> > If I transfer from the Polycom though, it seems to work.
>> >
>> > In resume:
>> > I cannot transfer from A leg. B leg doesn't seem to know its way around
>> and looses it. To demonstrate, I transfer B leg from A leg to extension
>> 5000.
>> > http://pastebin.freeswitch.org/19893
>> >
>> > I tried Googleing the error code and didn't find anything particularly
>> related to my issue.
>> >
>> > Any hint would be greatly appreciated. To make sure, I downgraded to
>> the stable version.
>> >
>> > Thanks a bunch, as always.
>> > 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 shishko69 at gmail.com  Sat Sep 15 14:08:14 2012
From: shishko69 at gmail.com (Shishko)
Date: Sat, 15 Sep 2012 12:08:14 +0200
Subject: [Freeswitch-users] Problem with SIP INVITE
In-Reply-To: <CAKDfjgeqED7CJ_y7ATxvKzL4S=PWAKk5zmD=8cdSOeVO+4ds7A@mail.gmail.com>
References: <5052EBDB.3020904@gmail.com>
	<CAKDfjgeqED7CJ_y7ATxvKzL4S=PWAKk5zmD=8cdSOeVO+4ds7A@mail.gmail.com>
Message-ID: <CAHT_yt8QaUsQDr8b5a0y9Dh+JQZJgHVN=fAg6TBqSzmycSHPeg@mail.gmail.com>

Thanks, Kristian.

On Fri, Sep 14, 2012 at 3:32 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> With late-negotiation the offer/answer model still applies.  They
> should be sending you an SDP in the ACK to your 200 OK.  They aren't.
> Their stuff is broken.
>
> On Fri, Sep 14, 2012 at 4:33 AM, Shishko <shishko69 at gmail.com> wrote:
>> Hi,
>>
>> one of my ITSP is using Cisco 2811 (IOS 12.4) as H.323 to SIP gateway
>> for SIP trunk to my Freeswitch.
>> I noticed that it sends INVITE message without SDP, and then Freeswitch
>> terminates the call.
>> I tried using 3pcc parameter as "proxy" and as "true", but no avail.
>>
>> Is this expected behavior, and could I fix it somehow?
>>
>> Here's SIP trace:
>>
>> ------------------------------------------------------------------------
>> recv 784 bytes from udp/[192.168.117.8]:51358 at 08:06:51.013018:
>> ------------------------------------------------------------------------
>>     INVITE sip:100965461 at 192.168.117.9:5060 SIP/2.0
>>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
>>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     To: <sip:100965461 at 192.168.117.9>
>>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     Supported: 100rel,timer,replaces
>>     Min-SE:  1800
>>     Cisco-Guid: 2649593691-218042625-2288844833-673556762
>>     User-Agent: Cisco-SIPGateway/IOS-12.x
>>     Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>>     CSeq: 101 INVITE
>>     Max-Forwards: 70
>>     Remote-Party-ID:
>> <sip:xxxxxxxxxxxx at 192.168.117.8>;party=calling;screen=yes;privacy=off
>>     Timestamp: 1347610034
>>     Contact: <sip:xxxxxxxxxxxx at 192.168.117.8:5060>
>>     Expires: 180
>>     Allow-Events: telephone-event
>>
>> ------------------------------------------------------------------------
>> send 372 bytes to udp/[192.168.117.8]:5060 at 08:06:51.013018:
>> ------------------------------------------------------------------------
>>     SIP/2.0 100 Trying
>>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
>>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     To: <sip:100965461 at 192.168.117.9>
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     CSeq: 101 INVITE
>>     Timestamp: 1347610034 0.000000
>>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>>     Content-Length: 0
>>
>> ------------------------------------------------------------------------
>> send 889 bytes to udp/[192.168.117.8]:5060 at 08:06:51.036020:
>> ------------------------------------------------------------------------
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK721016
>>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     CSeq: 101 INVITE
>>     Contact: <sip:mod_sofia at 192.168.117.9:5060>
>>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>>     Accept: application/sdp
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>     Supported: timer, precondition, path, replaces
>>     Allow-Events: talk, hold, conference, refer
>>     Min-SE: 1800
>>     Content-Type: application/sdp
>>     Content-Disposition: session
>>     Content-Length: 205
>>
>>     v=0
>>     o=FreeSWITCH 1347586709 1347586710 IN IP4 192.168.117.9
>>     s=FreeSWITCH
>>     c=IN IP4 192.168.117.9
>>     t=0 0
>>     m=audio 23302 RTP/AVP 8 0 101 13
>>     a=rtpmap:101 telephone-event/8000
>>     a=fmtp:101 0-16
>>     a=ptime:20
>> ------------------------------------------------------------------------
>> recv 364 bytes from udp/[192.168.117.8]:51358 at 08:06:51.041020:
>> ------------------------------------------------------------------------
>>     ACK sip:mod_sofia at 192.168.117.9:5060 SIP/2.0
>>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK731155
>>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     Max-Forwards: 70
>>     CSeq: 101 ACK
>>     Content-Length: 0
>>
>> ------------------------------------------------------------------------
>> send 645 bytes to udp/[192.168.117.8]:5060 at 08:06:51.041020:
>> ------------------------------------------------------------------------
>>     BYE sip:xxxxxxxxxxxx at 192.168.117.8:5060 SIP/2.0
>>     Via: SIP/2.0/UDP 192.168.117.9;rport;branch=z9hG4bKvBN2ZB9DZKZ6c
>>     Max-Forwards: 70
>>     From: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>>     To: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     CSeq: 33468941 BYE
>>     Contact: <sip:mod_sofia at 192.168.117.9:5060>
>>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>     Supported: timer, precondition, path, replaces
>>     Reason: SIP;cause=488;text="No answer to offer"
>>     Content-Length: 0
>>
>> ------------------------------------------------------------------------
>> recv 426 bytes from udp/[192.168.117.8]:51358 at 08:06:51.042020:
>> ------------------------------------------------------------------------
>>     BYE sip:mod_sofia at 192.168.117.9:5060 SIP/2.0
>>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
>>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     User-Agent: Cisco-SIPGateway/IOS-12.x
>>     Max-Forwards: 70
>>     Timestamp: 1347610034
>>     CSeq: 102 BYE
>>     Content-Length: 0
>>
>> ------------------------------------------------------------------------
>> send 387 bytes to udp/[192.168.117.8]:5060 at 08:06:51.042020:
>> ------------------------------------------------------------------------
>>     SIP/2.0 100 Trying
>>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
>>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     CSeq: 102 BYE
>>     Timestamp: 1347610034 0.000000
>>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>>     Content-Length: 0
>>
>> ------------------------------------------------------------------------
>> send 510 bytes to udp/[192.168.117.8]:5060 at 08:06:51.043021:
>> ------------------------------------------------------------------------
>>     SIP/2.0 480 Temporarily Unavailable
>>     Via: SIP/2.0/UDP  192.168.117.8:5060;branch=z9hG4bK741AD7
>>     From: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     To: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     CSeq: 102 BYE
>>     User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120907T231757Z~c2893801cb
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>     Supported: timer, precondition, path, replaces
>>     Content-Length: 0
>>
>> ------------------------------------------------------------------------
>> recv 363 bytes from udp/[192.168.117.8]:5060 at 08:06:51.046021:
>> ------------------------------------------------------------------------
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP 192.168.117.9;rport;branch=z9hG4bKvBN2ZB9DZKZ6c
>>     From: <sip:100965461 at 192.168.117.9>;tag=vUg243Ng67UHD
>>     To: <sip:xxxxxxxxxxxx at 192.168.117.8>;tag=21811124-711
>>     Date: Fri, 14 Sep 2012 08:07:14 GMT
>>     Call-ID: 6DE7A86-FD7A11E1-859AE929-E0E81CE1 at 192.168.117.8
>>     Server: Cisco-SIPGateway/IOS-12.x
>>     Content-Length: 0
>>     CSeq: 33468941 BYE
>>
>> ------------------------------------------------------------------------
>>
>> Thanks,
>>
>> Denis
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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


From mi.ke at null.net  Sat Sep 15 15:23:39 2012
From: mi.ke at null.net (Mi Ke)
Date: Sat, 15 Sep 2012 07:23:39 -0400
Subject: [Freeswitch-users] Variable continue on fail - ORIGINATOR_CANCEL
Message-ID: <20120915112339.6810@gmx.com>

Also, you may have noticed ORIGINATOR_CANCEL among other reject codes assigned to fail_on_single_reject in our dp. Logically it shouldn't be there as all these reject codes suppose to come from B-leg. However, when leg A disconnects first and bridge is still trying next route on Leg B, we were getting the following error:

 [ERR] switch_core_io.c:812 sofia/external/legA_calling_id at legA_calling_ip has received a bad frame with no codec!

 After few experiments we found that adding OC to FOSR var takes above error away - not sure this is correct approach, but it worked.

 Cheers,
 Mike




----- Original Message -----
From: Kirill Sysoev
Sent: 09/13/12 05:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Variable continue on fail

 Hi all!

 Really helpful and interesting answers.
 I can confirm that Vitaliy's lua script alreafy works like a charm for me.
fail_on_single_reject variable wasn't even noticed by me, while I was looking for a solution.
 It seems like it exists exactly for such situations...

 Thank you again!

 Regards,

 13.09.2012 18:19, Avi Marcus ?????:
 Can someone update the wiki http://wiki.freeswitch.org/wiki/Variable_continue_on_fail  on this? 
 I haven't quite figure out how to use this var.

 Thanks,
-Avi

 On Thu, Sep 13, 2012 at 4:58 PM, Mi Ke < mi.ke at null.net > wrote:
Hi Kirill,

 We use the following construction:

 <extension name="ex1">
 <condition field="destination_number" expression="your_dest_num_here">
 <action application="set" data="bridge_answer_timeout=120"/>
 <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,CALL_REJECTED,RECOVERY_ON_TIMER_EXPIRE"/>
 <action application="set" data="fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL"/> <---this prevents further failover in appropriate cases
 <action application="set" data="bridge_early_media=true"/>
 <action application="set" data="hangup_after_bridge=true"/>
 <action application="lua" data="get_dial_string.lua"/>
 <action application="bridge" data="${out_dial_string}"/>
 <anti-action application="pre_answer"/>
 <anti-action application="playback" data="/usr/local/freeswitch/recordings/number_incorrect.mp3"/>
 <anti-action application="hangup" data="INVALID_NUMBER_FORMAT"/>
 </condition>
 </extension>

 where get_dial_string app returns corresponding dialpeers as [dp1_params]sofia/external/dp1|[dp2_params]sofia/external/dp2|[dp3_params]sofia/external/dp3

 It works without any issues.

 FreeSWITCH Version 1.2.0-rc2+git~20120815T215213Z~c6d7417aec, CentOS 6.2 x64

 HTH,
 Mike

----- Original Message -----
From: Kirill Sysoev
Sent: 09/13/12 02:13 PM
Subject: Re: [Freeswitch-users] Variable continue on fail

 Hi David, Vitaly! Thank you for your answers. Will try to implement your advices. Best, Kirill 13.09.2012 14:52, ??????? ??????? ?????: > Hi, Kirill! > > I solved it by adding lua-script after "bridge" application in my > dialplan: > > endpoint_disposition = session:getVariable("endpoint_disposition") > last_bridge_hangup_cause = > session:getVariable("last_bridge_hangup_cause") > if (last_bridge_hangup_cause == "USER_BUSY" and endpoint_disposition > == "EARLY MEDIA" ) then > session:hangup(); > end > > You can see these variables by adding "info" application in the > corresponding extension. > > 12.09.2012 17:53, iam ?????: >> Hi Vitaly! >> >> Did you ever solve this problem? >> I have the same continue_on_fail behaviour. >> >> http://jira.freeswitch.org/browse/FS-4232  >> >> Vitaly added a comment - 26/Jun/12 3:14 AM >> Sorry for the long absence, but the problem is still actual for me. >> I replaced in my dialplan continue_on_fail to "true", but it's not give >> desired effect. >> May be some additional variables can help me? >> >> Regards, >> Kirill >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7462051p7582782.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 

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

 
 

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

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


 _________________________________________________________________________ Professional FreeSWITCH Consulting Services:  consulting at freeswitch.org http://www.freeswitchsolutions.com     Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com  FreeSWITCH-users mailing list  FreeSWITCH-users at lists.freeswitch.org http://lists.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  Sat Sep 15 18:24:57 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 15 Sep 2012 07:24:57 -0700 (PDT)
Subject: [Freeswitch-users] Problem with originated calls hanging
	in	CS_SOFT_EXEC state
In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FDF1F1@BLUPRD0711MB413.namprd07.prod.outlook.com>
References: <CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>
	<16C9AF65-B0CC-458E-8CDC-62B7BC733261@vocalspace.com>
	<CAKbxfG8vaTvAKfTUNMV+u=jm=6isqu2AVT=DHbVWcBzkbCfmpQ@mail.gmail.com>
	<87A42A1F-F4E3-40B0-88C7-5EFEE0AD328C@vocalspace.com>
	<CAKbxfG8xSExEuwBitxjHBH_HVuVofzc2MRnT96B+DTqq4ygN9A@mail.gmail.com>
	<20120904190858.cc516138@mail.tritonwest.net>
	<CAKbxfG9UL4zh1w5xTLoxqBzEHBJ94MRbY+6hvJu21Ze21ZCqsQ@mail.gmail.com>
	<6D0900A6-2881-47EF-8FEB-CC7354EA786E@vocalspace.com>
	<20120904211433.ad092146@mail.tritonwest.net>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FDF1F1@BLUPRD0711MB413.namprd07.prod.outlook.com>
Message-ID: <1347719097459-7582881.post@n2.nabble.com>

I made a small correction to the code which was causing the reported problem
here. The only requirement is that for every instance of ManagedSession you
call destroy when done with it.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-originated-calls-hanging-in-CS-SOFT-EXEC-state-tp7582377p7582881.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gavin.henry at gmail.com  Sun Sep 16 03:38:45 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sun, 16 Sep 2012 00:38:45 +0100
Subject: [Freeswitch-users] List of "status" statuses in a PRESENCE_IN
	event?
In-Reply-To: <CAA8_NKD0ZPvn-bRWzPxcd240V1-QbpvXNhVVLcsQDCbBiPXFgw@mail.gmail.com>
References: <CAA8_NKD0ZPvn-bRWzPxcd240V1-QbpvXNhVVLcsQDCbBiPXFgw@mail.gmail.com>
Message-ID: <CAA8_NKDY_bFft+Av-30ijfDoydgmHWsNnedgJ6iQWeDBKCGhZA@mail.gmail.com>

>
> Is there a list of what the "status" can be? Above has user-agent:
> Registered(UDP) but status: Unregistered. Is status if someone is
> subscribed to this presence info, like a BLF?
>
> How can I tell if a phone is online via a PRESENCE_IN?
>
> Re the "status" values, I've seen:
>
> CS_ROUTING == Ringing
> CS_HANGUP == HangUp :-)
> answered == Answered :-)
>
> I've also seen:
>
> status: Registered(TCP-NAT)
>
> status: Unregistered but user-agent saying Registered(TCP-NAT) too.
>
> How can I check the endpoint is available, i.e. which field and value
> drives a BLF being green, red and flashing? (I know that's obviously
> done via a SIP NOTIFY)
>

I'll try reading the source code to see if there is a list.

Thanks.

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


From andrew at cassidywebservices.co.uk  Sun Sep 16 12:04:15 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 16 Sep 2012 09:04:15 +0100
Subject: [Freeswitch-users] Kazoo Review
In-Reply-To: <CAHbF-TVQoDyw_gYqJcLB8vefdicX5_P+rBN2z4OCGCMhaU2jPA@mail.gmail.com>
References: <34429093.post@talk.nabble.com>
	<CAKzWOxXfHkTjQF+9sQUG2sXhCtpO_H=88GWjp1dGb3KEo+TjcQ@mail.gmail.com>
	<CAHbF-TVQoDyw_gYqJcLB8vefdicX5_P+rBN2z4OCGCMhaU2jPA@mail.gmail.com>
Message-ID: <CAMbW9oLKRZ_xom0wsyvcR2WDpK5Du_yckHKVN5WqyM2odtTYEA@mail.gmail.com>

I've got an account, but not tried it. I spend too much time building
custom systems!

On 14 September 2012 07:29, Tamurlang Clan <tamurlangclan at gmail.com> wrote:

> Sorry but thanks Micheal.
>
>
> On Fri, Sep 14, 2012 at 3:06 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> You are better off talking to the 2600hz community on this one. There are
>> a scant handful of people here who have used Kazoo but if you get on the
>> 2600hz group list you'll get to the target demographic you are seeking.
>>
>> -MC
>>
>>
>> On Thu, Sep 13, 2012 at 12:57 PM, TamurlangClan <TamurlangClan at gmail.com>wrote:
>>
>>>
>>> Hi,
>>>
>>> I'm thinking to use Kazoo by 2600HZ for effectively mainting PBX and
>>> other
>>> services. Has anyone used it? Anyone willing to give feedback on how
>>> effective it is? Thanks.
>>> --
>>> View this message in context:
>>> http://old.nabble.com/Kazoo-Review-tp34429093p34429093.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
>>>
>>
>>
>>
>> --
>> 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
>
>


-- 
*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 lists at kavun.ch  Mon Sep 17 02:24:01 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 16 Sep 2012 18:24:01 -0400
Subject: [Freeswitch-users] 481 Call Leg/Transaction Does Not Exist
In-Reply-To: <CANBhyPBjH_--apWHb5aGAkNy61Mhnmx6AK-i5H1Zesx352pUTQ@mail.gmail.com>
References: <5D5FD0B1-34C0-4C06-AD20-735E3ACE8803@kavun.ch>
	<99D9E8DC-8621-47E4-8F25-44F7B33001B2@kavun.ch>
	<CANBhyPAD7kuZ8_e8KoWSxvGpeaEh=oNkqWn89MNVGMqYMFzb2Q@mail.gmail.com>
	<CANBhyPBjH_--apWHb5aGAkNy61Mhnmx6AK-i5H1Zesx352pUTQ@mail.gmail.com>
Message-ID: <63E50EF2-0B87-4309-AE89-03BD7C6BED1C@kavun.ch>

> Hi Anton,

Good to know that my frustration is shared, I feel better. :P
I actually solved my issue.
For some reason, mod_sofia rewrites the domain name with the IP address of my machine, which causes the call to never reach the destination the transfer was intended to. I am guessing that the domain name = IP address is pulled from vars.xml? I am running my FS in multi tenant mode, even though I am the only user on it.

It seems like the 481 error never breaks the transfer. I don't know why it is there and why it is triggered? Some input from the developers would be good. The Yealink still sends a 481 even with a successful transfer.

Coincidentally, my attended transfer issues may have been solved too!

Now the question is, how do we make mod_sofia create new sessions with the proper domain name?

Best,
E

On Sep 15, 2012, at 3:11 AM, Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:

> So the question is: why the hack FS sends BYE to Yealink to tear down the session when this must do the UA which is performing the call transfer with REFER method. Obviously, Yealink responds with 481 to this kind of BYE, becauce it has already killed the dialog.
> 
> 2012/9/15 Anton Kvashenkin <anton.jugatsu at gmail.com>
> Emrah, can you please paste the successful dump of call when you transfer from Polycom. I observe strange bahaviour when FS sends BYE to Yealink to shutdown the dialog and Yealink also kills the dialog with it's BYE.
> 
> 
> 2012/9/15 Emrah <lists at kavun.ch>
> Hi all,
> 
> I have tried to play with the refer extension and now have my calls land in the public context when they are transferred?
> This is the second extension in my dialplan:
> <extension name="refer">
>  <condition field="${sip_refer_to}">
>         <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression>
>  </condition>
>  <condition field="${sip_refer_to}">
>         <expression><![CDATA[<sip:(.*)@(.*)>]]></expression>
>         <action application="set" data="refer_user=$1"/>
>         <action application="set" data="refer_domain=$2"/>
>         <action application="info"/>
>         <action application="bridge" data="sofia/${use_profile}/${refer_user}@${domain_name}"/>
>  </condition>
>     </extension>
> 
> Any help would be greatly appreciated.
> 
> All the best,
> Emrah
> On Sep 13, 2012, at 3:09 PM, Emrah <lists at kavun.ch> wrote:
> 
> > Hi all,
> >
> > I seem to be encountering the strangest problems.
> > I have been doing some more testing with my transfers and here is a recurring issue.
> >
> > I have 2 phones behind the same router. 1 Polycom VVX 1500 and 1 Yealink SIP-T38g
> > They are both registering on the same FS server which is on a remote public IP.
> > The Polycom has extension 20 and the yealink uses 21.
> > When I call the Polycom from the Yealink and perform a blind transfer from the Yealink, it fails with: 481 Call Leg/Transaction Does Not Exist.
> > If I transfer from the Polycom though, it seems to work.
> >
> > In resume:
> > I cannot transfer from A leg. B leg doesn't seem to know its way around and looses it. To demonstrate, I transfer B leg from A leg to extension 5000.
> > http://pastebin.freeswitch.org/19893
> >
> > I tried Googleing the error code and didn't find anything particularly related to my issue.
> >
> > Any hint would be greatly appreciated. To make sure, I downgraded to the stable version.
> >
> > Thanks a bunch, as always.
> > 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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.burlingame at me.com  Mon Sep 17 05:21:39 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sun, 16 Sep 2012 18:21:39 -0700
Subject: [Freeswitch-users] FreeSwitch not Processing calls (B-Leg) until
	restart
Message-ID: <213A77A7-64E9-4BA8-AF67-13E12FC86927@me.com>

I had an issue today not sure if anyone else has seen this and would be willing to share some information if you have - Today FreeSwitch stopped processing calls correctly the A-Leg of the call would come into FreeSwitch -  i see the notice for the new channel however no sip signaling was sent out for the B-Leg - I could not do any debug because fs_cli was locked up, force quoting fs_cli and restarting it yielded seeing the sip signaling however as soon as I type anything at the prompt fs_cli would lock again. The only resolution was to restart the FreeSwitch process and everything worked as expected again.

Has anyone else seen anything like this using git's from Sept 14th?


From lists at kavun.ch  Mon Sep 17 05:47:32 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 16 Sep 2012 21:47:32 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
Message-ID: <ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>

Guys, this is a recurring issue that can be reproduced on all the installs I've made.
I would really appreciate if we could eliminate the "in between the chair and laptop" possibility by having somebody else try.

Does your invalid prompt gets interrupted when you start dialing DTMFs?

Thanks!

On Sep 13, 2012, at 1:30 PM, Emrah <lists at kavun.ch> wrote:

> To clarify further:
> 
> The DTMFs are capture when the invalid prompt is played and only executed after the prompt ends, as if they were queued.
> 
> To reproduce:
> 
> Execute the play_and_get_digits app with an invalid prompt of something a little longer than 5 secondes, just to give you enough time to enter some digits.
> When the initial prompt plays, dial some digits that do not match your regex so that you end up in the invalid prompt.
> While the invalid prompt is playing, try dialing some DTMFs as if you had made a mistake and were dialing the correct sequence this time around.
> Even if you key "#", the invalid prompt will still be played till the end and all your digits queued for execution right after.
> 
> Thanks for testing,
> Emrah
> On Sep 13, 2012, at 4:36 AM, covici at ccs.covici.com wrote:
> 
>> I am not seeing this -- I can hit the dtmf while the prompt is still
>> playing.  When its playing the invalid prompt, what could you type
>> anyway -- it may not be listening yet.
>> 
>> Emrah <lists at kavun.ch> wrote:
>> 
>>> This happens with the stable version as well. It seems to be limited to the play_and_get_digits app though.
>>> 
>>> Best,
>>> Emrah
>>> On Sep 13, 2012, at 2:34 AM, Emrah <lists at kavun.ch> wrote:
>>> 
>>>> Thanks Michael. I will check out the stable version and let you know.
>>>> 
>>>> Regards,
>>>> Emrah
>>>> On Sep 11, 2012, at 8:02 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>> 
>>>>> I do not see this behavior on 1.2.stable branch. When I press a key during the invalid prompt it breaks out of playback immediately and attempts to process whatever digit(s) I input. I tested with x5000 sample IVR:
>>>>> Dial 5000
>>>>> Press 7, wait for "invalid entry" message to start playing
>>>>> Press 3 while she's saying "that was an invalid entry"
>>>>> After a few seconds MOH comes on.
>>>>> 
>>>>> -MC
>>>>> 
>>>>> On Tue, Sep 11, 2012 at 12:43 PM, Emrah <lists at kavun.ch> wrote:
>>>>> Just wanted to drop a quick note about an issue I just stumbled upon.
>>>>> 
>>>>> When using play_and_get_digits, you are still able to input DTMFs when the invalid-entry prompt is played, but the message won't be interrupted and your DTMF input won't be executed until the prompt is fully played.
>>>>> 
>>>>> Can someone look into this and confirm?
>>>>> 
>>>>> 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
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
> 



From anthony.minessale at gmail.com  Mon Sep 17 05:56:21 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 16 Sep 2012 20:56:21 -0500
Subject: [Freeswitch-users] FreeSwitch not Processing calls (B-Leg)
	until restart
In-Reply-To: <213A77A7-64E9-4BA8-AF67-13E12FC86927@me.com>
References: <213A77A7-64E9-4BA8-AF67-13E12FC86927@me.com>
Message-ID: <CAKbxfG-eD63rN7f=D=-tz8nUDdWb0A=L7k6_NDOYnFOWZRzjvg@mail.gmail.com>

recourse in that situation would be to try

fs_cli -x "<cmd>"

if all else fails, gcore the fs process with the gcore command and get a bt
just like a crash.


On Sun, Sep 16, 2012 at 8:21 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> I had an issue today not sure if anyone else has seen this and would be
> willing to share some information if you have - Today FreeSwitch stopped
> processing calls correctly the A-Leg of the call would come into FreeSwitch
> -  i see the notice for the new channel however no sip signaling was sent
> out for the B-Leg - I could not do any debug because fs_cli was locked up,
> force quoting fs_cli and restarting it yielded seeing the sip signaling
> however as soon as I type anything at the prompt fs_cli would lock again.
> The only resolution was to restart the FreeSwitch process and everything
> worked as expected again.
>
> Has anyone else seen anything like this using git's from Sept 14th?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike.burlingame at me.com  Mon Sep 17 05:59:03 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sun, 16 Sep 2012 18:59:03 -0700
Subject: [Freeswitch-users] FreeSwitch not Processing calls
 (B-Leg)	until restart
In-Reply-To: <CAKbxfG-eD63rN7f=D=-tz8nUDdWb0A=L7k6_NDOYnFOWZRzjvg@mail.gmail.com>
References: <213A77A7-64E9-4BA8-AF67-13E12FC86927@me.com>
	<CAKbxfG-eD63rN7f=D=-tz8nUDdWb0A=L7k6_NDOYnFOWZRzjvg@mail.gmail.com>
Message-ID: <AD377EE5-7A7F-438A-9E0F-ADE2936D9DFE@me.com>

Noted 

Thank you
On Sep 16, 2012, at 6:56 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> recourse in that situation would be to try
> 
> fs_cli -x "<cmd>"
> 
> if all else fails, gcore the fs process with the gcore command and get a bt just like a crash.
> 
> 
> On Sun, Sep 16, 2012 at 8:21 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> I had an issue today not sure if anyone else has seen this and would be willing to share some information if you have - Today FreeSwitch stopped processing calls correctly the A-Leg of the call would come into FreeSwitch -  i see the notice for the new channel however no sip signaling was sent out for the B-Leg - I could not do any debug because fs_cli was locked up, force quoting fs_cli and restarting it yielded seeing the sip signaling however as soon as I type anything at the prompt fs_cli would lock again. The only resolution was to restart the FreeSwitch process and everything worked as expected again.
> 
> Has anyone else seen anything like this using git's from Sept 14th?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 georg at riseup.net  Sun Sep 16 22:23:23 2012
From: georg at riseup.net (georg at riseup.net)
Date: Sun, 16 Sep 2012 20:23:23 +0200
Subject: [Freeswitch-users] mod_snom + mod_fifo: Turning the light off
Message-ID: <16f3ca27ae8adf2908b318fb0b304acf.squirrel@fulvetta.riseup.net>

Hi all,

I'm using FS in a small office. I've setup a small queue with mod_fifo.
Now I woud like let my users know, when someone enters the queue and
they're still talking. I've used mod_snom for this, which works great.
Hoever, I'm not able to turn the light off after the call is completed.


<extension name="kunden-queue">
    <condition field="destination_number" expression="^(666)$">
        <!-- <key> <light> <label> <user> <host> <profile> <action_name>
<action> -->
        <action application="eval" data="${snom_bind_key(2 on x kunden-1
192.168.1.93 internal x x)}"/>
        <action application="fifo" data="fifo_kunden at 192.168.1.93 in undef
/data/usr/local/freeswitch/sounds/kunden-queue.raw" />
        <action application="transfer" data="6666 XML public" />
        <action application="eval" data="${snom_bind_key(2 off x kunden-1
192.168.1.93 internal x x)}"/>
    </condition>
</extension>

It seems, that the dialplan is not processed anymore, after the caller
gets into the queue.

Any hints for this?

Thanks,
Georg



From Alexander.Haugg at c4b.de  Mon Sep 17 10:13:32 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Mon, 17 Sep 2012 06:13:32 +0000
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>

Hi MC,

thank you for the answer.
To your question, all call legs in this scenario (outgoing or incoming direction) are connected over a sip trunk of a pbx.

Scenario:
My CTI
  |  A
  |  |       event socket
  V  |
Freeswitch         Sip Trunk
<-                           PBX
->

My Client control (CTI) the call legs over the event socket interface and the call legs are only legs over the sip trunk to or from the PBX.

After some tests i have found a possible solution:

-          Leg A and Leg B are bridged (all legs get the flag park_after_bridge = true)

-          For Consultation i park Leg B and transfer this Leg to Moh in my default context of my dialplan

-          i originate Leg C and bridge this Leg with Leg A (Leg C have the flag park_after_bridge = true too) Consultation is comlete now.

-          with the principle same think i can toggle Leg A <-> Leg B and Leg A <-> Leg C.

What is your think for this solution? I have tested this on the FS CLI and it works.

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins
Gesendet: Freitag, 14. September 2012 19:13
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Consultation Call via event_socket interface

Hi Alex,

Welcome to the FreeSWITCH mail list!

First question for you: what kind of telephone are you using? The reason I ask is that this kind of function is trivially achieved with a good hard phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys. If you can use a hard phone w/ multiple line keys then you don't even need to mess with the dialplan, uuid_bridge, etc.

-MC
On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi All,

I'm new on the mailing list.
I have a problem with a call scenario.

-          Channel A and channel B are bridged (A is my own channel and B is my calling partner)

-          Now i set channel B on hold with the command "uuid_hold xxx" and create a new channel to C with the command:

bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip',origination_caller_id_number=num at ip}sofia/external/num at ip  &park

this works correctly, the partner C answer and the channel is established.

-          Now the Problem:

I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false.

Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment.
Is there a general problem in my plan to do that?
Is it a better plan to do this over the dialplan?
The next step in this scenario is to toggle the connection A -> B and A -> C.

Thanks for your help!
Nice regards,
Alex

_________________________________________________________________________
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 miha at softnet.si  Mon Sep 17 13:24:48 2012
From: miha at softnet.si (Miha)
Date: Mon, 17 Sep 2012 11:24:48 +0200
Subject: [Freeswitch-users] Phones lossing connection
Message-ID: <5056EC60.3030807@softnet.si>

Hi,

my Fs version is 1.2 rc2.

I have about 820 phones registered on it. I noticed that after one 
mounth of production 20 phones behind nat lost connection and I needed 
to restart all 20 phones (I was unable to get them from outside, also 
calls from inside FS were not working).

On think is that the Switch could be causing problems but I did not have 
to restart switch, I just restarted phones...

What do you thing, could this be FS problem?

p.s.: other phones, which are not behind nat are working as they should.

Regards,
MIha


From jaganthoutam at gmail.com  Mon Sep 17 13:36:19 2012
From: jaganthoutam at gmail.com (Jagadish Thoutam)
Date: Mon, 17 Sep 2012 15:06:19 +0530
Subject: [Freeswitch-users] issue with web server to handle XML CDRs
Message-ID: <CAHqn9HFfo9JiB5mz747Pg18q5hPZXgfV6cnroyQF_SqBkcepzA@mail.gmail.com>

Hi all,

   There is the issue with xml_cdr for me when i try to call <param
name="url" value="http://localhost/cgi-bin/cdr.pl"/> its not working, i
have load the mod_xml_cdr on cli, and my cdr.pl file is working well when
try it manually and also have executable permission for cdr.pl file, please
some one tell me what i am missing here.



Thanks

Jagadish
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From jmesquita at freeswitch.org  Mon Sep 17 16:19:41 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 17 Sep 2012 09:19:41 -0300
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
Message-ID: <CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>

>From personal experience, I believe that how you described is the right way
to do it. The only caveat is that you will have to add variables do the
channels so you can properly track what is going on in the cdrs . If you
don't process cdrs, then it is all good. Careful with pickup and such too...
On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de> wrote:

>  Hi MC,****
>
> ** **
>
> thank you for the answer.****
>
> To your question, all call legs in this scenario (outgoing or incoming
> direction) are connected over a sip trunk of a pbx.****
>
> ** **
>
> Scenario:****
>
> My CTI****
>
>   |  A****
>
>   |  |       event socket****
>
>   V  |****
>
> Freeswitch         Sip Trunk****
>
> <-                           PBX****
>
> ->                           ****
>
> ** **
>
> My Client control (CTI) the call legs over the event socket interface and
> the call legs are only legs over the sip trunk to or from the PBX.****
>
> ** **
>
> After some tests i have found a possible solution:****
>
> **-          **Leg A and Leg B are bridged (all legs get the flag
> park_after_bridge = true)****
>
> **-          **For Consultation i park Leg B and transfer this Leg to Moh
> in my default context of my dialplan****
>
> **-          **i originate Leg C and bridge this Leg with Leg A (Leg C
> have the flag park_after_bridge = true too) Consultation is comlete now.**
> **
>
> **-          **with the principle same think i can toggle Leg A <-> Leg B
> and Leg A <-> Leg C.****
>
> ** **
>
> What is your think for this solution? I have tested this on the FS CLI and
> it works.****
>
> ** **
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Michael
> Collins
> *Gesendet:* Freitag, 14. September 2012 19:13
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] Consultation Call via event_socket
> interface****
>
> ** **
>
> Hi Alex,
>
> Welcome to the FreeSWITCH mail list!
>
> First question for you: what kind of telephone are you using? The reason I
> ask is that this kind of function is trivially achieved with a good hard
> phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys.
> If you can use a hard phone w/ multiple line keys then you don't even need
> to mess with the dialplan, uuid_bridge, etc.
>
> -MC****
>
> On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
> wrote:****
>
> Hi All,****
>
>  ****
>
> I?m new on the mailing list.****
>
> I have a problem with a call scenario.****
>
> -          Channel A and channel B are bridged (A is my own channel and B
> is my calling partner)****
>
> -          Now i set channel B on hold with the command ?uuid_hold xxx?
> and create a new channel to C with the command:****
>
> bgapi originate
> {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip
> ',origination_caller_id_number=num at ip}sofia/external/num at ip  &park****
>
> this works correctly, the partner C answer and the channel is established.
> ****
>
> -          Now the Problem:****
>
> I try to bridge the channel a to channel c with the uuid_bridge command,
> now the channel b will hangup, why? The variable hangup_after_bridge is by
> default false.****
>
> Other problem: channel A can hear the voice of channel C but not speak
> with him, channel C can hear and speak. But this problem is not the
> important think at the moment.****
>
> Is there a general problem in my plan to do that?****
>
> Is it a better plan to do this over the dialplan?****
>
> The next step in this scenario is to toggle the connection A -> B and A ->
> C.****
>
>  ****
>
> Thanks for your help!****
>
> Nice regards,****
>
> 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****
>
>
>
>
> --
> 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 garmt.noname at gmail.com  Mon Sep 17 17:06:12 2012
From: garmt.noname at gmail.com (grmt)
Date: Mon, 17 Sep 2012 15:06:12 +0200
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
	<ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
Message-ID: <002a01cd94d5$3c0a3820$b41ea860$@gmail.com>

I didn't spend time (yet) to reproduce your issue, but you may want to give
us some more information:

What kind of DTMF (inband /outband - RFC2833) are you using, what is your
client?
Are you testing locally or through a provider?
Maybe share your dialplan/script ...
Are you using text-to-speech?







From vipkilla at gmail.com  Mon Sep 17 17:24:31 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 17 Sep 2012 09:24:31 -0400
Subject: [Freeswitch-users] no audio from portaudio in conference
Message-ID: <CAC-LwPPEw641dyzuWU6OwTQYTqfNfsoTGr9xfV-51EGydc1ZuQ@mail.gmail.com>

I have audio playing into the MIC (capture) on my sound card. The
device is "/dev/dsp"
I know there is audio being played into the sound card because when I run:
sox -t ossdsp -w -s /dev/dsp  ./tmp.wav
and listen to tmp.wav, I hear the audio.
When I call a conference using "/dev/dsp" no audio is being played
from the capture into the conference. If I use freeswitch-1.06,
portaudio seems to work. I'd like to use the latest FS version for
many reasons. Any ideas what has changed in mod_portaudio? Maybe
something broke? It doesn't seem like many people use this module and
may not be maintained or bugs may not be submitted for it.
I'll do any debugging necessary and any advice would be appreciated.
Thanks.


From Rob.Moore at Aeriandi.com  Mon Sep 17 18:30:35 2012
From: Rob.Moore at Aeriandi.com (Rob Moore)
Date: Mon, 17 Sep 2012 14:30:35 +0000
Subject: [Freeswitch-users] Altering From Header in SIP Invite
In-Reply-To: <CAKzWOxUjc=LSQ8S5Hh6wd=ofNHg8-n=Uf2QebPNBkOwdNw_yJg@mail.gmail.com>
References: <49C5FCA19A8A114493EBAACA42FE5899105C87C2@1AERDCEXCHMBX1.AER.AERCO.local>
	<49C5FCA19A8A114493EBAACA42FE5899105CA323@1AERDCEXCHMBX1.AER.AERCO.local>
	<CAKzWOxUjc=LSQ8S5Hh6wd=ofNHg8-n=Uf2QebPNBkOwdNw_yJg@mail.gmail.com>
Message-ID: <49C5FCA19A8A114493EBAACA42FE5899105CC020@1AERDCEXCHMBX1.AER.AERCO.local>

Oh certainly,

It certainly wouldn't be a lynch job on them, it was just a very unusual setup they asked for .
The provider was British Telecom (BT) on their IP Exchange platform.

Can I just add a page or do I have to get a log on for the wiki? I'm currently writing up an article for our Tech blog at work that will give a summary of the tests passed during the week and why. Would be hand to post a link through to that for anyone looking to pass the BT IOT in the future.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 14 September 2012 17:39
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Altering From Header in SIP Invite

Without throwing the provider under the bus it might be good to document this as a "known issue with a workaround." We would just need to add a provider page and link it here:
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

Thanks!
-MC
On Fri, Sep 14, 2012 at 4:46 AM, Rob Moore <Rob.Moore at aeriandi.com<mailto:Rob.Moore at aeriandi.com>> wrote:
Hi Everyone,

For those who are interested I found a way around this issue. I'm not sure if this is the correct way to produce this result but it worked.

The problem I had was that I could set the Freephone number I wished to  present but the P asserted ID would overwrite this with the other standard number I was attempting to send.

So to resolve the issue I had to disable all CID, use effective_caller_id_number to present my Freephone number (this adds the number to your from header) then I used sip_h_ to add the P-Asserted-Identity manually.

                                <action application="set" data="effective_caller_id_number=+448000000262<tel:%2B448000000262>"/>
                                <action application="export" data="sip_cid_type=none"/>
                                <action application="export" data="sip_h_P-Asserted-Identity= <sip:+445600000262<tel:%2B445600000262>@fixsipbizzle.org<http://fixsipbizzle.org>> "/>

Hope this helps anyone else who ever has to provide this unusual setup.

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 Rob Moore
Sent: 11 September 2012 19:11
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Altering From Header in SIP Invite

Hi All,

I'm having a little trouble with 'presentation numbers' with a new provider I'm in IOT with this week.

I'm trying to recreate the following Invite as the calls pass through our dialplan to this provider but there are issues with trying to get a different CLI into the From and P-Asserted-Identity headers.

When presenting a Freephone number (for example) we need to still present the regular number that would be used by this extension in the P-Asserted-Identity whilst passing the number we wish to have presented in the From header.

Currently we are not using Gateways so we cannot resort to using  <param name="caller-id-in-from" value="true"/> (although I expect this won't do what we need in this case) so I've looked at altering the channel variables  sip_from_user,sip_full_from and sip_full_uri using set and export dial plan apps but none of these seem to have any effect so I guess these variables must be read only.

I'm sure this must be simple, but can't for the life of me work out what I need to do.

Below is an extract from an example header from the provider I am trying to recreate, I've also added a copy of the Dialplan extension I am using to test.
If someone can tell me what I'm getting wrong I would really appreciate it.

Thanks

Rob


INVITE sip:+445600005262 at primarysip.barfoo.com;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: +445600000262<tel:%2B445600000262> <sip:+445600005262 at primarysip.barfoo.com;user=phone>
From: <sip:+448000655262 at uk.foo.bar.2.net;user=phone>;tag=12544
P-Asserted-Identity: <sip:+445600655262 at uk.sdin.bt.net>
Call-ID: 1347372978-13100 at mgc-uk-998.n2<mailto:1347372978-13100 at mgc-uk-998.n2>
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE


DIAL Plan: (attempting +445600655262<tel:%2B445600655262> in P Asserted-Id and +448000655262<tel:%2B448000655262> in from) have commented out some things that I have tried.)
If you are worried about [sip_h_X-Gateway=4003:4] this is added to help our SBC forward calls to our different providers.

   <extension name="TC 25">
     <condition field="destination_number" expression="^10125$">
                                <action application="set" data="effective_caller_id_name=_undef_"/>
                                <action application="set" data="effective_caller_id_number=+445600005262<tel:%2B445600005262> "/>
                                <action application="set" data="dtmf_type=rfc2833"/>
                                <action application="set" data="sip_cid_type=pid"/>
                                <!--<action application="export" data=" sip_from_user =+448000655262<tel:%2B448000655262>"/> -->
                                <!-- <action application="export" data="sip_h_from=<sip:+448000655262 at primarysip.barfoo.com>;tag=${sip_from_tag} "/> -->
                                <!--  <action application="export" data="sip_h_P-Asserted-Identity= +445600005262<tel:%2B445600005262> <sip:+445600655262 at 172.17.7.4> "/> -->
                                <action application="info"/>
                  <action application="bridge" data="[sip_h_X-Gateway=4003:4]sofia/internal/+442920001199 at pstn.tel<mailto:sofia/internal/+442920001199 at pstn.tel>" />
     </condition>
   </extension>


Rob Moore
Telephony Systems Infrastructure Manager
Aeriandi
Aeriandi Ltd, Prama House, Banbury Road, Oxford, OX27HT
E: rob.moore at aeriandi.com<mailto:rob.moore at aeriandi.com>
W: www.aeriandi.com<http://www.aeriandi.com/>
M: +44 (0)7766 838040<tel:%2B44%20%280%297766%20838040>
T: +44 (0) 845 108 0308

[Description: Description: Description: Description: Description: Description: Description: Description: cid:image002.png at 01CC9E0C.20153A40]


_________________________________________________________________________
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 lists at kavun.ch  Mon Sep 17 18:53:05 2012
From: lists at kavun.ch (Emrah)
Date: Mon, 17 Sep 2012 10:53:05 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <002a01cd94d5$3c0a3820$b41ea860$@gmail.com>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
	<ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
	<002a01cd94d5$3c0a3820$b41ea860$@gmail.com>
Message-ID: <0A200C25-FFE2-4D2F-84C8-7DD3B53007EB@kavun.ch>

Hey there,

I use RFC2833 directly from my hardware phone to FS. The issue is not client dependent and happens with any UA.
Here is the dialplan sequence that triggers to the play_and_get_digits

<action application="answer" />
<action application="sleep" data="1000" />
<action application="play_and_get_digits" data="4 5 3 10000 # /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/pls-enter-conf-room-number.wav /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/conf-number-invalid-contact-conf-administrator.wav conf_num ^30\d{2}$ 3000" />

Any idea would be greatly appreciated.

Best,
Emrah
On Sep 17, 2012, at 9:06 AM, grmt <garmt.noname at gmail.com> wrote:

> I didn't spend time (yet) to reproduce your issue, but you may want to give
> us some more information:
> 
> What kind of DTMF (inband /outband - RFC2833) are you using, what is your
> client?
> Are you testing locally or through a provider?
> Maybe share your dialplan/script ...
> Are you using text-to-speech?
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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.burlingame at me.com  Mon Sep 17 19:00:12 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Mon, 17 Sep 2012 08:00:12 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
	<CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
Message-ID: <694713AF-A904-4574-9B63-C476C0BE15E3@me.com>

PaeTec and Global Crossing both feel that we should not be including the require based on the RFC posted below - I have read over the RFC and tend to agree based on my limited knowledge of the timer RFC

Based on what they are saying because the invite from them state that timers are supported that we should remember this and not be putting in a require

------------------
From carriers 
-------------------

Please see below?section 8.1 of RFC 4028
 
From RFC4028:
 
   The proxy MUST remember, for the duration of the transaction, whether
   the request contained the Supported header field with the value
   'timer'.  If the request did not contain a Supported header field
   with the value 'timer', the proxy *MAY* insert a Require header field
   with the value 'timer' into the request.  However, this is*NOT
   RECOMMENDED*.  This allows the proxy to insist on a session timer for
   the session.  This header field is not needed if a Supported header
   field was in the request; in this case, the proxy would already be
   sure the session timer can be used for the session.
 
 
Section 7.1 of RFC 4028
 
A UAC that supports the session timer extension defined here MUST
   include a Supported header field in each request (except ACK),
   listing the option tag 'timer' [2].  It MUST do so even if the UAC is
   not requesting usage of the session timer for this session.
 
   The UAC MAY include a Require header field in the request with the
   value 'timer' to indicate that the UAS must support the session timer
   to participate in the session.  This does not mean that the UAC is
   requiring the UAS to perform the refreshes, only that it is requiring
   the UAS to support the extension.  In addition, the UAC MAY include a
   Proxy-Require header field in the request with the value 'timer' to
   indicate that proxies must support the session timer in order to
   correctly process the request.  However, usage of either Require or
   Proxy-Require by the UAC is NOT RECOMMENDED.  They are not needed,
   since the extension works even when only the UAC supports the
   extension.  The Supported header field containing 'timer' MUST still
   be included, even if the Require or Proxy-Require header fields are
   present containing 'timer'
 

Sent from my iPhone 4S

On Sep 14, 2012, at 5:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> It's Wong to not accept it. IIRC...there is a better case for why it should be there than for it not being there.  It's a rather dumb thing for some of these commercial switches to even care about....I don't want to set a precedence here that I will perpetually change our sip stack and chase after every interop case... Make sure the param is set, I believe it disables it completely going against the rfc but I can't check till Monday.  This is part of Sofia not FS itself.
> 
> On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
>> Thanks that seem to help the high session timer value however the require: timer is still present 
>> 
>> Invite from B-Leg --> FreeSwitch
>> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>>    ------------------------------------------------------------------------
>>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>>    Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>>    Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>>    Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
>>    Max-Forwards: 68
>>    Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>>    From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>>    CSeq: 219015274 INVITE
>>    Expires: 180
>>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>>    Organization: MetaSwitch
>>    Supported: 100rel, resource-priority
>>    Content-Length: 193
>>    Content-Type: application/sdp
>>    
>>    v=0
>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>    s=-
>>    c=IN IP4 Carrier_IP
>>    t=0 0
>>    m=audio 29864 RTP/AVP 18 0 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=ptime:20
>>    a=silenceSupp:off - - - -
>> 
>> 
>> 
>> Invite from FreeSwitch --> A-Leg
>> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>>    ------------------------------------------------------------------------
>>    INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>>    Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>>    Route: <sip:ULC;lr=on>
>>    Max-Forwards: 47
>>    From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>>    CSeq: 33497250 INVITE
>>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>>    User-Agent: FreeSwitch
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>    Require: timer
>>    Supported: timer, precondition, path, replaces
>>    Session-Expires: 3600;refresher=uas
>>    Min-SE: 120
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 193
>>    P-hint: outbound
>>    X-FS-Support: update_display,send_info
>>    
>>    v=0
>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>    s=-
>>    c=IN IP4 Carrier_IP
>>    t=0 0
>>    m=audio 29864 RTP/AVP 18 0 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=ptime:20
>>    a=silenceSupp:off - - - -
>> 
>> 
>> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> Mike, try comma separating the values instead of using two sets of {}:
>>> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>> 
>>> -MC
>>> 
>>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> I have that set in the bridge command however does not seem to be working as expected.
>>>> 
>>>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>>> 
>>>> 
>>>> 
>>>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>> 
>>>>> set {sip_require_timer=false} in your outbound calls or globally 
>>>>> 
>>>>> 
>>>>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>>> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
>>>>>> 
>>>>>> The ULC are stating we need to only have it in our supported and do not pass them a require.
>>>>>> 
>>>>>> Re-Invite from B-Leg to FS
>>>>>>    ------------------------------------------------------------------------
>>>>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>>>>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>>>>    Content-Length: 217
>>>>>>    Content-Type: application/sdp
>>>>>>    CSeq: 33480808 INVITE
>>>>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>>>>    Max-Forwards: 92
>>>>>>    Session-Expires: 3600;refresher=uas
>>>>>>    Supported: timer
>>>>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>>>>    User-Agent: FreeSwitch
>>>>>> 
>>>>>>    v=0
>>>>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>>>>    s=Media Server
>>>>>>    c=IN IP4 B-LEG_IP
>>>>>>    t=0 0
>>>>>>    m=audio 51246 RTP/AVP 0 101
>>>>>>    a=rtpmap:0 PCMU/8000
>>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>>    a=fmtp:101 0-15
>>>>>>    a=ptime:20
>>>>>>    ------------------------------------------------------------------------
>>>>>> 
>>>>>> Re-Invite from FS to A-Leg
>>>>>>   ------------------------------------------------------------------------
>>>>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>>>>    Max-Forwards: 97
>>>>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>>>>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>>>>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>>>>    CSeq: 33480811 INVITE
>>>>>>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>>>>    User-Agent: FreeSwitch
>>>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>>>>>    Require: timer
>>>>>>    Supported: timer, precondition, path, replaces
>>>>>>    Session-Expires: 64800;refresher=uas
>>>>>>    Min-SE: 64800
>>>>>>    Content-Type: application/sdp
>>>>>>    Content-Disposition: session
>>>>>>    Content-Length: 235
>>>>>>    X-FS-Support: update_display,send_info
>>>>>> 
>>>>>>    v=0
>>>>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>>>>    s=Media Server
>>>>>>    c=IN IP4 B-LEG_IP
>>>>>>    t=0 0
>>>>>>    m=audio 51246 RTP/AVP 0 101
>>>>>>    a=rtpmap:0 PCMU/8000
>>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>>    a=fmtp:101 0-15
>>>>>>    a=ptime:20
>>>>>>    ------------------------------------------------------------------------
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>> 
>>> 
>>> 
>>> -- 
>>> 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
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From anthony.minessale at gmail.com  Mon Sep 17 19:11:25 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Sep 2012 10:11:25 -0500
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <694713AF-A904-4574-9B63-C476C0BE15E3@me.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
	<CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
	<694713AF-A904-4574-9B63-C476C0BE15E3@me.com>
Message-ID: <CAKbxfG8oGeJkboDL69j=W0a0ozXKCX-MHPBjaNs+1Sm6Pw0UgQ@mail.gmail.com>

Just because it says its not recommended does not mean it says you should
intentionally drop a call because of it.
If you can make it work by turning off in the var then that is your
solution.  As I have already said, there are several real-world situations
where it breaks things going either way hence our solution was to make it
work both ways.


On Mon, Sep 17, 2012 at 10:00 AM, Mike Burlingame <mike.burlingame at me.com>wrote:

> PaeTec and Global Crossing both feel that we should not be including the
> require based on the RFC posted below - I have read over the RFC and tend
> to agree based on my limited knowledge of the timer RFC
>
> Based on what they are saying because the invite from them state that
> timers are supported that we should remember this and not be putting in a
> require
>
> ------------------
> From carriers
> -------------------
>
> Please see below?section 8.1 of RFC 4028****
>
>
>
> From RFC4028:****
>
>
>
>    The proxy MUST remember, for the duration of the transaction, whether**
> **
>
>    the request contained the Supported header field with the value****
>
>    'timer'.  If the request did not contain a Supported header field****
>
>    with the value 'timer', the proxy *MAY* insert a Require header field**
> **
>
>    with the value 'timer' into the request.  However, this is*NOT****
>
>    RECOMMENDED*.  This allows the proxy to insist on a session timer for**
> **
>
>    the session.  This header field is not needed if a Supported header****
>
>    field was in the request; in this case, the proxy would already be****
>
>    sure the session timer can be used for the session.****
>
>
>
>
>
> Section 7.1 of RFC 4028****
>
>
>
> A UAC that supports the session timer extension defined here MUST****
>
>    include a Supported header field in each request (except ACK),****
>
>    listing the option tag 'timer' [2<http://tools.ietf.org/html/rfc4028#ref-2>].
> It MUST do so even if the UAC is****
>
>    not requesting usage of the session timer for this session.****
>
>
>
>    The UAC MAY include a Require header field in the request with the****
>
>    value 'timer' to indicate that the UAS must support the session timer**
> **
>
>    to participate in the session.  This does not mean that the UAC is****
>
>    requiring the UAS to perform the refreshes, only that it is requiring**
> **
>
>    the UAS to support the extension.  In addition, the UAC MAY include a**
> **
>
>    Proxy-Require header field in the request with the value 'timer' to****
>
>    indicate that proxies must support the session timer in order to****
>
>    correctly process the request.  However, usage of either Require or****
>
>    Proxy-Require by the UAC is NOT RECOMMENDED.  They are not needed,****
>
>    since the extension works even when only the UAC supports the****
>
>    extension.  The Supported header field containing 'timer' MUST still***
> *
>
>    be included, even if the Require or Proxy-Require header fields are****
>
>    present containing 'timer'****
>
>
>
> Sent from my iPhone 4S
>
> On Sep 14, 2012, at 5:29 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> It's Wong to not accept it. IIRC...there is a better case for why it
> should be there than for it not being there.  It's a rather dumb thing for
> some of these commercial switches to even care about....I don't want to set
> a precedence here that I will perpetually change our sip stack and chase
> after every interop case... Make sure the param is set, I believe it
> disables it completely going against the rfc but I can't check till
> Monday.  This is part of Sofia not FS itself.
> On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
>
>> Thanks that seem to help the high session timer value however the
>> require: timer is still present
>>
>> Invite from B-Leg --> FreeSwitch
>> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>>
>>  ------------------------------------------------------------------------
>>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>>    Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>>    Via: SIP/2.0/UDP
>> Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>>    Allow-Events: message-summary, refer, dialog, line-seize, presence,
>> call-info, as-feature-event
>>    Max-Forwards: 68
>>    Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>>    From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>>    CSeq: 219015274 INVITE
>>    Expires: 180
>>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>>    Organization: MetaSwitch
>>    Supported: 100rel, resource-priority
>>    Content-Length: 193
>>    Content-Type: application/sdp
>>
>>    v=0
>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>    s=-
>>    c=IN IP4 Carrier_IP
>>    t=0 0
>>    m=audio 29864 RTP/AVP 18 0 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=ptime:20
>>    a=silenceSupp:off - - - -
>>
>>
>>
>> Invite from FreeSwitch --> A-Leg
>> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>>
>>  ------------------------------------------------------------------------
>>    INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>>    Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>>    Route: <sip:ULC;lr=on>
>>    Max-Forwards: 47
>>    From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>>    CSeq: 33497250 INVITE
>>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>>    User-Agent: FreeSwitch
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>    Require: timer
>>    Supported: timer, precondition, path, replaces
>>    Session-Expires: 3600;refresher=uas
>>    Min-SE: 120
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 193
>>    P-hint: outbound
>>    X-FS-Support: update_display,send_info
>>
>>    v=0
>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>    s=-
>>    c=IN IP4 Carrier_IP
>>    t=0 0
>>    m=audio 29864 RTP/AVP 18 0 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=ptime:20
>>    a=silenceSupp:off - - - -
>>
>>
>> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> Mike, try comma separating the values instead of using two sets of {}:
>>
>> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>
>> -MC
>>
>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>>
>>> I have that set in the bridge command however does not seem to be
>>> working as expected.
>>>
>>>
>>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>>
>>>
>>>
>>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>> set {sip_require_timer=false} in your outbound calls or globally
>>>
>>>
>>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com
>>> > wrote:
>>>
>>>> it seems if I get an re-invite from the B-Leg FS add's requires timer
>>>> and changes the session timer to a high value to the re-invite going to the
>>>> A-Leg come to find out Acme Packets at our ULC's do not like this and send
>>>> us back a 420 Bad Extension and the call is disconnected with Reason:
>>>> Q.850;cause=127;text="INTERWORKING"
>>>>
>>>> The ULC are stating we need to only have it in our supported and do not
>>>> pass them a require.
>>>>
>>>> Re-Invite from B-Leg to FS
>>>>
>>>>  ------------------------------------------------------------------------
>>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>>    Via: SIP/2.0/UDP
>>>> B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>>    Content-Length: 217
>>>>    Content-Type: application/sdp
>>>>    CSeq: 33480808 INVITE
>>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>>    Max-Forwards: 92
>>>>    Session-Expires: 3600;refresher=uas
>>>>    Supported: timer
>>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>>    User-Agent: FreeSwitch
>>>>
>>>>    v=0
>>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>>    s=Media Server
>>>>    c=IN IP4 B-LEG_IP
>>>>    t=0 0
>>>>    m=audio 51246 RTP/AVP 0 101
>>>>    a=rtpmap:0 PCMU/8000
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-15
>>>>    a=ptime:20
>>>>
>>>>  ------------------------------------------------------------------------
>>>>
>>>> Re-Invite from FS to A-Leg
>>>>
>>>> ------------------------------------------------------------------------
>>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>>    Max-Forwards: 97
>>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy
>>>> :5060;user=phone>;tag=c9Favaa53XFXB
>>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER
>>>> :5060;user=phone>;tag=SDd626401-gK095bbb72
>>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>>    CSeq: 33480811 INVITE
>>>>    Contact: <
>>>> sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>>    User-Agent: FreeSwitch
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> NOTIFY
>>>>    Require: timer
>>>>    Supported: timer, precondition, path, replaces
>>>>    Session-Expires: 64800;refresher=uas
>>>>    Min-SE: 64800
>>>>    Content-Type: application/sdp
>>>>    Content-Disposition: session
>>>>    Content-Length: 235
>>>>    X-FS-Support: update_display,send_info
>>>>
>>>>    v=0
>>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>>    s=Media Server
>>>>    c=IN IP4 B-LEG_IP
>>>>    t=0 0
>>>>    m=audio 51246 RTP/AVP 0 101
>>>>    a=rtpmap:0 PCMU/8000
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-15
>>>>    a=ptime:20
>>>>
>>>>  ------------------------------------------------------------------------
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>> http://www.ClueCon.com <http://www.cluecon.com/>
>> http://www.OSTAG.org <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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike.burlingame at me.com  Mon Sep 17 19:17:39 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Mon, 17 Sep 2012 08:17:39 -0700
Subject: [Freeswitch-users] Require Timer on re-invite from FS to A-LEG
In-Reply-To: <CAKbxfG8oGeJkboDL69j=W0a0ozXKCX-MHPBjaNs+1Sm6Pw0UgQ@mail.gmail.com>
References: <67ADD323-2463-40DD-88D7-96844B123C50@me.com>
	<CAKbxfG-MTxeR_FWz8apDBgd73MmD-yMTSCPzTfTqM8kivf4suQ@mail.gmail.com>
	<80F8DD09-CF05-4AC1-BCAE-0692B8A4EB85@me.com>
	<CAKzWOxW2VXzxCu49F7Y0osx_MnkXFssUEEXUVd6qqhRZBpTO9A@mail.gmail.com>
	<58797397-E56A-4B70-AF67-3203EBC0109E@me.com>
	<CAKbxfG96vhj5KoUdJ+uNxrh2Jb++ye3j-MDt8VGaHUzFRk_r2g@mail.gmail.com>
	<694713AF-A904-4574-9B63-C476C0BE15E3@me.com>
	<CAKbxfG8oGeJkboDL69j=W0a0ozXKCX-MHPBjaNs+1Sm6Pw0UgQ@mail.gmail.com>
Message-ID: <1ACCFCB3-A8BA-4D13-8B5C-C2772087DF19@me.com>

I agree they should not be dropping the call - was just passing on the info to complete out the thread as well as show it seems to be losing battle with the carriers to try to get them to change

Sent from my iPhone 4S

On Sep 17, 2012, at 8:11 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Just because it says its not recommended does not mean it says you should intentionally drop a call because of it.
> If you can make it work by turning off in the var then that is your solution.  As I have already said, there are several real-world situations where it breaks things going either way hence our solution was to make it work both ways.
> 
> 
> On Mon, Sep 17, 2012 at 10:00 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> PaeTec and Global Crossing both feel that we should not be including the require based on the RFC posted below - I have read over the RFC and tend to agree based on my limited knowledge of the timer RFC
>> 
>> Based on what they are saying because the invite from them state that timers are supported that we should remember this and not be putting in a require
>> 
>> ------------------
>> From carriers 
>> -------------------
>> 
>> Please see below?section 8.1 of RFC 4028
>>  
>> From RFC4028:
>>  
>>    The proxy MUST remember, for the duration of the transaction, whether
>>    the request contained the Supported header field with the value
>>    'timer'.  If the request did not contain a Supported header field
>>    with the value 'timer', the proxy *MAY* insert a Require header field
>>    with the value 'timer' into the request.  However, this is*NOT
>>    RECOMMENDED*.  This allows the proxy to insist on a session timer for
>>    the session.  This header field is not needed if a Supported header
>>    field was in the request; in this case, the proxy would already be
>>    sure the session timer can be used for the session.
>>  
>>  
>> Section 7.1 of RFC 4028
>>  
>> A UAC that supports the session timer extension defined here MUST
>>    include a Supported header field in each request (except ACK),
>>    listing the option tag 'timer' [2].  It MUST do so even if the UAC is
>>    not requesting usage of the session timer for this session.
>>  
>>    The UAC MAY include a Require header field in the request with the
>>    value 'timer' to indicate that the UAS must support the session timer
>>    to participate in the session.  This does not mean that the UAC is
>>    requiring the UAS to perform the refreshes, only that it is requiring
>>    the UAS to support the extension.  In addition, the UAC MAY include a
>>    Proxy-Require header field in the request with the value 'timer' to
>>    indicate that proxies must support the session timer in order to
>>    correctly process the request.  However, usage of either Require or
>>    Proxy-Require by the UAC is NOT RECOMMENDED.  They are not needed,
>>    since the extension works even when only the UAC supports the
>>    extension.  The Supported header field containing 'timer' MUST still
>>    be included, even if the Require or Proxy-Require header fields are
>>    present containing 'timer'
>>  
>> 
>> Sent from my iPhone 4S
>> 
>> On Sep 14, 2012, at 5:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> It's Wong to not accept it. IIRC...there is a better case for why it should be there than for it not being there.  It's a rather dumb thing for some of these commercial switches to even care about....I don't want to set a precedence here that I will perpetually change our sip stack and chase after every interop case... Make sure the param is set, I believe it disables it completely going against the rfc but I can't check till Monday.  This is part of Sofia not FS itself.
>>> 
>>> On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
>>>> Thanks that seem to help the high session timer value however the require: timer is still present 
>>>> 
>>>> Invite from B-Leg --> FreeSwitch
>>>> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>>>>    ------------------------------------------------------------------------
>>>>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>>>>    Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>>>>    Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>>>>    Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
>>>>    Max-Forwards: 68
>>>>    Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>>>>    From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>>>>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>>>>    CSeq: 219015274 INVITE
>>>>    Expires: 180
>>>>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>>>>    Organization: MetaSwitch
>>>>    Supported: 100rel, resource-priority
>>>>    Content-Length: 193
>>>>    Content-Type: application/sdp
>>>>    
>>>>    v=0
>>>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>>>    s=-
>>>>    c=IN IP4 Carrier_IP
>>>>    t=0 0
>>>>    m=audio 29864 RTP/AVP 18 0 101
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=ptime:20
>>>>    a=silenceSupp:off - - - -
>>>> 
>>>> 
>>>> 
>>>> Invite from FreeSwitch --> A-Leg
>>>> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>>>>    ------------------------------------------------------------------------
>>>>    INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>>>>    Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>>>>    Route: <sip:ULC;lr=on>
>>>>    Max-Forwards: 47
>>>>    From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>>>>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>>>>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>>>>    CSeq: 33497250 INVITE
>>>>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>>>>    User-Agent: FreeSwitch
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>>>    Require: timer
>>>>    Supported: timer, precondition, path, replaces
>>>>    Session-Expires: 3600;refresher=uas
>>>>    Min-SE: 120
>>>>    Content-Type: application/sdp
>>>>    Content-Disposition: session
>>>>    Content-Length: 193
>>>>    P-hint: outbound
>>>>    X-FS-Support: update_display,send_info
>>>>    
>>>>    v=0
>>>>    o=- 3341834898 3341834898 IN IP4 Carrier_IP
>>>>    s=-
>>>>    c=IN IP4 Carrier_IP
>>>>    t=0 0
>>>>    m=audio 29864 RTP/AVP 18 0 101
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=ptime:20
>>>>    a=silenceSupp:off - - - -
>>>> 
>>>> 
>>>> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>> 
>>>>> Mike, try comma separating the values instead of using two sets of {}:
>>>>> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>>>> 
>>>>> -MC
>>>>> 
>>>>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>>> I have that set in the bridge command however does not seem to be working as expected.
>>>>>> 
>>>>>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>>>> 
>>>>>>> set {sip_require_timer=false} in your outbound calls or globally 
>>>>>>> 
>>>>>>> 
>>>>>>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>>>>> it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"
>>>>>>>> 
>>>>>>>> The ULC are stating we need to only have it in our supported and do not pass them a require.
>>>>>>>> 
>>>>>>>> Re-Invite from B-Leg to FS
>>>>>>>>    ------------------------------------------------------------------------
>>>>>>>>    INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>>>>>>    Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>>>>>>    Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>>>>>>    Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>>>>>>    Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>>>>>>    Content-Length: 217
>>>>>>>>    Content-Type: application/sdp
>>>>>>>>    CSeq: 33480808 INVITE
>>>>>>>>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>>>>>>    Max-Forwards: 92
>>>>>>>>    Session-Expires: 3600;refresher=uas
>>>>>>>>    Supported: timer
>>>>>>>>    To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>>>>>>    User-Agent: FreeSwitch
>>>>>>>> 
>>>>>>>>    v=0
>>>>>>>>    o=- 3308986892 0 IN IP4 B-LEG_IP
>>>>>>>>    s=Media Server
>>>>>>>>    c=IN IP4 B-LEG_IP
>>>>>>>>    t=0 0
>>>>>>>>    m=audio 51246 RTP/AVP 0 101
>>>>>>>>    a=rtpmap:0 PCMU/8000
>>>>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>>>>    a=fmtp:101 0-15
>>>>>>>>    a=ptime:20
>>>>>>>>    ------------------------------------------------------------------------
>>>>>>>> 
>>>>>>>> Re-Invite from FS to A-Leg
>>>>>>>>   ------------------------------------------------------------------------
>>>>>>>>    INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>>>>>>    Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>>>>>>    Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>>>>>>    Max-Forwards: 97
>>>>>>>>    From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>>>>>>>>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>>>>>>>>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>>>>>>    CSeq: 33480811 INVITE
>>>>>>>>    Contact: <sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>>>>>>    User-Agent: FreeSwitch
>>>>>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>>>>>>>>    Require: timer
>>>>>>>>    Supported: timer, precondition, path, replaces
>>>>>>>>    Session-Expires: 64800;refresher=uas
>>>>>>>>    Min-SE: 64800
>>>>>>>>    Content-Type: application/sdp
>>>>>>>>    Content-Disposition: session
>>>>>>>>    Content-Length: 235
>>>>>>>>    X-FS-Support: update_display,send_info
>>>>>>>> 
>>>>>>>>    v=0
>>>>>>>>    o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>>>>>>    s=Media Server
>>>>>>>>    c=IN IP4 B-LEG_IP
>>>>>>>>    t=0 0
>>>>>>>>    m=audio 51246 RTP/AVP 0 101
>>>>>>>>    a=rtpmap:0 PCMU/8000
>>>>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>>>>    a=fmtp:101 0-15
>>>>>>>>    a=ptime:20
>>>>>>>>    ------------------------------------------------------------------------
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>> 
>>>>> 
>>>>> 
>>>>> -- 
>>>>> 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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From christian.loeschenkohl at xpirio.com  Mon Sep 17 19:28:10 2012
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 17 Sep 2012 17:28:10 +0200
Subject: [Freeswitch-users] t.38 re-invite results in 407 Proxy
	Authentication Required
Message-ID: <5057418A.6010006@xpirio.com>

hello fs users

anybody have similar problems than me?

symptom is that within a t.38 re-invite freeswitch responds with 407 Proxy Authentication Required
wasn't the case in e.g. 1.2.0-rc2
now with 1.2.2+git~20120911T185917Z~3fd1a57902 it's like described

was this an intended changed. if yes - why?
many sip ua devices only respond with an ACK instead of an re-invite with user credentials.

br

-- 
Ing. Christian L?schenkohl
Senior VoIP Engineer
Head of VoIP Engineering
Head of R&D

T +43 5 77 11 - 1000
F +43 5 77 11 - 1002
E christian.loeschenkohl at xpirio.com
W www.xpirio.com

xpirio Telekommunikation & Service GmbH

Gerichtsstand Klagenfurt FN297465z
Lakeside B04 | 9020 Klagenfurt | Austria

An ESS Group company


From msc at freeswitch.org  Mon Sep 17 19:35:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Sep 2012 08:35:43 -0700
Subject: [Freeswitch-users] You don't need to save a greeting to
 override your existing one in Voicemail
In-Reply-To: <F07A307C-BE0A-43C7-8C88-0B7C411C7D8E@kavun.ch>
References: <F07A307C-BE0A-43C7-8C88-0B7C411C7D8E@kavun.ch>
Message-ID: <CAKzWOxUrcurCUfgrVnyp2jWqv6OJQfc=hRnXX7RKayLD4=cH5g@mail.gmail.com>

Emrah,

I think this is doable with some programming in mod_voicemail.c although I
don't believe it's a high priority at the moment. However, you can manually
emulate this behavior by recording a new greeting number that is different
from your current, active greeting. For example, if your current greeting
is 1, then record greeting 2, make sure it's satisfactory, then set 2 as
your active greeting. It is basically an extra step, but at least it
doesn't require any coding changes and it will get the job done.

-MC

On Fri, Sep 14, 2012 at 2:30 PM, Emrah <lists at kavun.ch> wrote:

> It used to be the case with Asterisk as well, now it has changed.
> Basically, if you have a greeting in place and want to record a new one?
> And you suddenly cough in the middle of your recording? Than get
> interrupted by your boss right after you've pressed #? You just hang up the
> phone and think that your existing greeting hasn't been altered because you
> didn't press 2 to save your re-recorded greeting?
> Well that's wrong.
>
> Your existing greeting gets overridden as soon as you start the recording.
> Even if you hang up the phone in the middle of your message, it'll still be
> recorded and played as your greeting.
>
> I don't think that this calls for a workaround or more studying of the
> app. I am using the default settings on this and think that it can be
> improved, so that you can review your greeting before committing it with a
> save action.
> Being able to listen to your existing greeting before re-recording it
> wouldn't be a bad add-on either.
>
> What is your take on this?
>
> 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 mitch.capper at gmail.com  Mon Sep 17 19:52:31 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 17 Sep 2012 08:52:31 -0700
Subject: [Freeswitch-users] no audio from portaudio in conference
In-Reply-To: <CAC-LwPPEw641dyzuWU6OwTQYTqfNfsoTGr9xfV-51EGydc1ZuQ@mail.gmail.com>
References: <CAC-LwPPEw641dyzuWU6OwTQYTqfNfsoTGr9xfV-51EGydc1ZuQ@mail.gmail.com>
Message-ID: <CALn233o_r-CTdMf0of-vwQ-+ro9aGf9jhCV+2rR+qWEru1J1MA@mail.gmail.com>

FSClient uses portaudio at its core and is built against 1.2 it even
does conferencing.   Does portaudio work for a normal call?   Is it
just the mic input not working or is the speakerphone output also not
working?

~Mitch

On Mon, Sep 17, 2012 at 6:24 AM, Vik Killa <vipkilla at gmail.com> wrote:
> I have audio playing into the MIC (capture) on my sound card. The
> device is "/dev/dsp"
> I know there is audio being played into the sound card because when I run:
> sox -t ossdsp -w -s /dev/dsp  ./tmp.wav
> and listen to tmp.wav, I hear the audio.
> When I call a conference using "/dev/dsp" no audio is being played
> from the capture into the conference. If I use freeswitch-1.06,
> portaudio seems to work. I'd like to use the latest FS version for
> many reasons. Any ideas what has changed in mod_portaudio? Maybe
> something broke? It doesn't seem like many people use this module and
> may not be maintained or bugs may not be submitted for it.
> I'll do any debugging necessary and any advice would be appreciated.
> 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 vipkilla at gmail.com  Mon Sep 17 20:00:10 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 17 Sep 2012 12:00:10 -0400
Subject: [Freeswitch-users] no audio from portaudio in conference
In-Reply-To: <CALn233o_r-CTdMf0of-vwQ-+ro9aGf9jhCV+2rR+qWEru1J1MA@mail.gmail.com>
References: <CAC-LwPPEw641dyzuWU6OwTQYTqfNfsoTGr9xfV-51EGydc1ZuQ@mail.gmail.com>
	<CALn233o_r-CTdMf0of-vwQ-+ro9aGf9jhCV+2rR+qWEru1J1MA@mail.gmail.com>
Message-ID: <CAC-LwPM5ikXpyjP90Z_fXPqZoD321gqvM-+-3kCwSXyWkWesRA@mail.gmail.com>

I haven't tested that.... I'm running on CentOS 5.6

On Mon, Sep 17, 2012 at 11:52 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
> FSClient uses portaudio at its core and is built against 1.2 it even
> does conferencing.   Does portaudio work for a normal call?   Is it
> just the mic input not working or is the speakerphone output also not
> working?
>
> ~Mitch


From mitch.capper at gmail.com  Mon Sep 17 20:11:24 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 17 Sep 2012 09:11:24 -0700
Subject: [Freeswitch-users] no audio from portaudio in conference
In-Reply-To: <CAC-LwPM5ikXpyjP90Z_fXPqZoD321gqvM-+-3kCwSXyWkWesRA@mail.gmail.com>
References: <CAC-LwPPEw641dyzuWU6OwTQYTqfNfsoTGr9xfV-51EGydc1ZuQ@mail.gmail.com>
	<CALn233o_r-CTdMf0of-vwQ-+ro9aGf9jhCV+2rR+qWEru1J1MA@mail.gmail.com>
	<CAC-LwPM5ikXpyjP90Z_fXPqZoD321gqvM-+-3kCwSXyWkWesRA@mail.gmail.com>
Message-ID: <CALn233oPfO5EfjTVVpoU2MGUvCgWzJZBG=mUWZhoR3-jumpKKg@mail.gmail.com>

Try that let us know:)

~Mitch

On Mon, Sep 17, 2012 at 9:00 AM, Vik Killa <vipkilla at gmail.com> wrote:
> I haven't tested that.... I'm running on CentOS 5.6
>
> On Mon, Sep 17, 2012 at 11:52 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
>> FSClient uses portaudio at its core and is built against 1.2 it even
>> does conferencing.   Does portaudio work for a normal call?   Is it
>> just the mic input not working or is the speakerphone output also not
>> working?
>>
>> ~Mitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Mon Sep 17 20:34:49 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Sep 2012 09:34:49 -0700
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <0A200C25-FFE2-4D2F-84C8-7DD3B53007EB@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
	<ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
	<002a01cd94d5$3c0a3820$b41ea860$@gmail.com>
	<0A200C25-FFE2-4D2F-84C8-7DD3B53007EB@kavun.ch>
Message-ID: <CAKzWOxX0ck=HWZzQp7sg0xbL9AZYRrrfJNuSKYbW0XMqHMoNHw@mail.gmail.com>

FTR, I was able to reproduce this behavior. If you haven't already done so
you can file a jira.
-MC

On Mon, Sep 17, 2012 at 7:53 AM, Emrah <lists at kavun.ch> wrote:

> Hey there,
>
> I use RFC2833 directly from my hardware phone to FS. The issue is not
> client dependent and happens with any UA.
> Here is the dialplan sequence that triggers to the play_and_get_digits
>
> <action application="answer" />
> <action application="sleep" data="1000" />
> <action application="play_and_get_digits" data="4 5 3 10000 #
> /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/pls-enter-conf-room-number.wav
> /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/conf-number-invalid-contact-conf-administrator.wav
> conf_num ^30\d{2}$ 3000" />
>
> Any idea would be greatly appreciated.
>
> Best,
> Emrah
> On Sep 17, 2012, at 9:06 AM, grmt <garmt.noname at gmail.com> wrote:
>
> > I didn't spend time (yet) to reproduce your issue, but you may want to
> give
> > us some more information:
> >
> > What kind of DTMF (inband /outband - RFC2833) are you using, what is your
> > client?
> > Are you testing locally or through a provider?
> > Maybe share your dialplan/script ...
> > Are you using text-to-speech?
> >
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Sep 17 20:37:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Sep 2012 09:37:17 -0700
Subject: [Freeswitch-users] issue with web server to handle XML CDRs
In-Reply-To: <CAHqn9HFfo9JiB5mz747Pg18q5hPZXgfV6cnroyQF_SqBkcepzA@mail.gmail.com>
References: <CAHqn9HFfo9JiB5mz747Pg18q5hPZXgfV6cnroyQF_SqBkcepzA@mail.gmail.com>
Message-ID: <CAKzWOxXz5aCKXBFt0a1wFysMUC6_wCzJCC=M9usJqKHPKfgXRw@mail.gmail.com>

Next step  is to watch the fs_cli to see if there are any errors. Also,
look at your web server access and error logs to see if the request is
making it there.
-MC

On Mon, Sep 17, 2012 at 2:36 AM, Jagadish Thoutam <jaganthoutam at gmail.com>wrote:

> Hi all,
>
>    There is the issue with xml_cdr for me when i try to call <param
> name="url" value="http://localhost/cgi-bin/cdr.pl"/> its not working, i
> have load the mod_xml_cdr on cli, and my cdr.pl file is working well when
> try it manually and also have executable permission for cdr.pl file,
> please some one tell me what i am missing here.
>
>
>
> Thanks
>
> Jagadish
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Sep 17 20:39:26 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Sep 2012 09:39:26 -0700
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
	<CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>
Message-ID: <CAKzWOxWFgXz5fnDUStz4q53x0C3vRdrR4LFPsY2c9u6f69a2AA@mail.gmail.com>

Or get a hard phone that has a hold button and at least two line keys.
-MC

On Mon, Sep 17, 2012 at 5:19 AM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:

> From personal experience, I believe that how you described is the right
> way to do it. The only caveat is that you will have to add variables do the
> channels so you can properly track what is going on in the cdrs . If you
> don't process cdrs, then it is all good. Careful with pickup and such too...
> On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de> wrote:
>
>>  Hi MC,****
>>
>> ** **
>>
>> thank you for the answer.****
>>
>> To your question, all call legs in this scenario (outgoing or incoming
>> direction) are connected over a sip trunk of a pbx.****
>>
>> ** **
>>
>> Scenario:****
>>
>> My CTI****
>>
>>   |  A****
>>
>>   |  |       event socket****
>>
>>   V  |****
>>
>> Freeswitch         Sip Trunk****
>>
>> <-                           PBX****
>>
>> ->                           ****
>>
>> ** **
>>
>> My Client control (CTI) the call legs over the event socket interface and
>> the call legs are only legs over the sip trunk to or from the PBX.****
>>
>> ** **
>>
>> After some tests i have found a possible solution:****
>>
>> **-          **Leg A and Leg B are bridged (all legs get the flag
>> park_after_bridge = true)****
>>
>> **-          **For Consultation i park Leg B and transfer this Leg to
>> Moh in my default context of my dialplan****
>>
>> **-          **i originate Leg C and bridge this Leg with Leg A (Leg C
>> have the flag park_after_bridge = true too) Consultation is comlete now.*
>> ***
>>
>> **-          **with the principle same think i can toggle Leg A <-> Leg
>> B and Leg A <-> Leg C.****
>>
>> ** **
>>
>> What is your think for this solution? I have tested this on the FS CLI
>> and it works.****
>>
>> ** **
>>
>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Michael
>> Collins
>> *Gesendet:* Freitag, 14. September 2012 19:13
>> *An:* FreeSWITCH Users Help
>> *Betreff:* Re: [Freeswitch-users] Consultation Call via event_socket
>> interface****
>>
>> ** **
>>
>> Hi Alex,
>>
>> Welcome to the FreeSWITCH mail list!
>>
>> First question for you: what kind of telephone are you using? The reason
>> I ask is that this kind of function is trivially achieved with a good hard
>> phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys.
>> If you can use a hard phone w/ multiple line keys then you don't even need
>> to mess with the dialplan, uuid_bridge, etc.
>>
>> -MC****
>>
>> On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
>> wrote:****
>>
>> Hi All,****
>>
>>  ****
>>
>> I?m new on the mailing list.****
>>
>> I have a problem with a call scenario.****
>>
>> -          Channel A and channel B are bridged (A is my own channel and
>> B is my calling partner)****
>>
>> -          Now i set channel B on hold with the command ?uuid_hold xxx?
>> and create a new channel to C with the command:****
>>
>> bgapi originate
>> {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip
>> ',origination_caller_id_number=num at ip}sofia/external/num at ip  &park****
>>
>> this works correctly, the partner C answer and the channel is established.
>> ****
>>
>> -          Now the Problem:****
>>
>> I try to bridge the channel a to channel c with the uuid_bridge command,
>> now the channel b will hangup, why? The variable hangup_after_bridge is by
>> default false.****
>>
>> Other problem: channel A can hear the voice of channel C but not speak
>> with him, channel C can hear and speak. But this problem is not the
>> important think at the moment.****
>>
>> Is there a general problem in my plan to do that?****
>>
>> Is it a better plan to do this over the dialplan?****
>>
>> The next step in this scenario is to toggle the connection A -> B and A
>> -> C.****
>>
>>  ****
>>
>> Thanks for your help!****
>>
>> Nice regards,****
>>
>> 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****
>>
>>
>>
>>
>> --
>> 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 marketing at cluecon.com  Mon Sep 17 21:42:59 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 17 Sep 2012 10:42:59 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVMy_iVviNsmBuGdwLUAb26z3wudUMk_KotMcaRnf+DZQ@mail.gmail.com>

Hello all,

It's been another productive week on the FreeSWITCH team. We are pleased to
let you know that we have officially tagged
<http://freeswitch.org/node/418>FreeSWITCH version 1.2.2 in the git
repo. Source tarballs are available in
the usual location <http://files.freeswitch.org>. Thanks to all those whose
efforts make more frequent releases a reality. It is much appreciated.

On last Wednesday's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_12>we enjoyed a
nice
Adhearsion <http://adhearsion.com/> presentation by Ben Langfeld and Ben
Klang. Adhearsion is a Ruby-based framework for developing telephony
applications. Ben and Ben discuss how Adhearsion works, why Ruby is cool
for building telephony apps, and why the Adhearsion guys love FreeSWITCH.
FreeSWITCH community members are invited to join the Adhearsion team at
AdhearsionConf <http://adhearsionconf.com> in Palo Alto, CA on October
20-21, 2012. Community members receive a special rate by using discount
code AHNLOVESFREESWITCH. Thanks to Ben and Ben for a great presentation
with cool slides<http://wiki.freeswitch.org/images/8/8e/FreeSWITCH_Weekly_CC_2012-09-12.pdf>
.

For the next few weeks we look forward to hearing from Daniel Pocock and
Scott Godin who will be telling us more about the
Repro<http://www.resiprocate.org/About_Repro>SIP proxy and the
ReSIProcate <http://www.resiprocate.org/Resip_Overview> SIP stack. For many
of us it will be our first look at a SIP proxy that does not have its roots
in the OpenSER project. We look forward to learning more on this
Wednesday's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19>
.

Have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From ZAlex at webley.com  Mon Sep 17 22:58:38 2012
From: ZAlex at webley.com (Alex Zarubin)
Date: Mon, 17 Sep 2012 13:58:38 -0500
Subject: [Freeswitch-users] SBC bridged call: DTMF delay
Message-ID: <74D0BA2985A4B04E8651FA01C8A58F3905E153FE@vs-win-ex01.corp.parusi.com>

Hi all,

 

FS in SBC mode, 'external' and 'internal' sip calls are bridged. DTMF
sequence '11234567890' gets passed with a delay, - 2 sec between the
last digit '0' is received and sent (see log below). There is a
consistent 500ms+ between 'Queue digit delay of 40ms' and 'Send start
packet' for the next digit.

Haven't found a way to fix it so far (wiki, user list, parameters such
as auto-rtp-bugs), help will be appreciated. CentOS 6.1, Xen, rfc2833.

 

Thank you.

 

Alex

 

 

2012-09-17 12:52:20.714917 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF
1:1120

2012-09-17 12:52:20.714917 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:20.774914 [DEBUG] switch_rtp.c:2420 Send start packet
for [1] ts=325320 dur=160/160/1120 seq=6880 lw=325320

2012-09-17 12:52:20.794915 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=325320 dur=320/320/1120 seq=6881 lw=325480

2012-09-17 12:52:20.814914 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=325320 dur=480/480/1120 seq=6882 lw=325640

2012-09-17 12:52:20.834915 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=325320 dur=640/640/1120 seq=6883 lw=325800

2012-09-17 12:52:20.874917 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=325320 dur=800/800/1120 seq=6884 lw=161

2012-09-17 12:52:20.894918 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=325320 dur=960/960/1120 seq=6885 lw=321

2012-09-17 12:52:20.914916 [DEBUG] switch_rtp.c:2323 Send end packet for
[1] ts=325320 dur=1120/1120/1120 seq=6886 lw=321

2012-09-17 12:52:20.914916 [DEBUG] switch_rtp.c:2323 Send end packet for
[1] ts=325320 dur=1120/1120/1120 seq=6887 lw=321

2012-09-17 12:52:20.914916 [DEBUG] switch_rtp.c:2323 Send end packet for
[1] ts=325320 dur=1120/1120/1120 seq=6888 lw=321

2012-09-17 12:52:20.914916 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:21.054920 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 1:960

2012-09-17 12:52:21.054920 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:21.454915 [DEBUG] switch_rtp.c:2420 Send start packet
for [1] ts=331880 dur=160/160/960 seq=6907 lw=331880

2012-09-17 12:52:21.474918 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=331880 dur=320/320/960 seq=6908 lw=332040

2012-09-17 12:52:21.494916 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 2:800

2012-09-17 12:52:21.494916 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=331880 dur=480/480/960 seq=6909 lw=332200

2012-09-17 12:52:21.494916 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:21.514915 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=331880 dur=640/640/960 seq=6910 lw=332360

2012-09-17 12:52:21.534920 [DEBUG] switch_rtp.c:2323 Send middle packet
for [1] ts=331880 dur=800/800/960 seq=6911 lw=332520

2012-09-17 12:52:21.554917 [DEBUG] switch_rtp.c:2323 Send end packet for
[1] ts=331880 dur=960/960/960 seq=6912 lw=332520

2012-09-17 12:52:21.554917 [DEBUG] switch_rtp.c:2323 Send end packet for
[1] ts=331880 dur=960/960/960 seq=6913 lw=332520

2012-09-17 12:52:21.554917 [DEBUG] switch_rtp.c:2323 Send end packet for
[1] ts=331880 dur=960/960/960 seq=6914 lw=332520

2012-09-17 12:52:21.554917 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:21.934916 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 3:800

2012-09-17 12:52:21.934916 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:22.094915 [DEBUG] switch_rtp.c:2420 Send start packet
for [2] ts=337000 dur=160/160/800 seq=6934 lw=337000

2012-09-17 12:52:22.114914 [DEBUG] switch_rtp.c:2323 Send middle packet
for [2] ts=337000 dur=320/320/800 seq=6935 lw=337160

2012-09-17 12:52:22.134920 [DEBUG] switch_rtp.c:2323 Send middle packet
for [2] ts=337000 dur=480/480/800 seq=6936 lw=337320

2012-09-17 12:52:22.154919 [DEBUG] switch_rtp.c:2323 Send middle packet
for [2] ts=337000 dur=640/640/800 seq=6937 lw=337480

2012-09-17 12:52:22.174917 [DEBUG] switch_rtp.c:2323 Send end packet for
[2] ts=337000 dur=800/800/800 seq=6938 lw=337480

2012-09-17 12:52:22.174917 [DEBUG] switch_rtp.c:2323 Send end packet for
[2] ts=337000 dur=800/800/800 seq=6939 lw=337480

2012-09-17 12:52:22.174917 [DEBUG] switch_rtp.c:2323 Send end packet for
[2] ts=337000 dur=800/800/800 seq=6940 lw=337480

2012-09-17 12:52:22.174917 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:22.554917 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 4:960

2012-09-17 12:52:22.554917 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:22.714915 [DEBUG] switch_rtp.c:2420 Send start packet
for [3] ts=341960 dur=160/160/800 seq=6959 lw=341960

2012-09-17 12:52:22.734917 [DEBUG] switch_rtp.c:2323 Send middle packet
for [3] ts=341960 dur=320/320/800 seq=6960 lw=342120

2012-09-17 12:52:22.754918 [DEBUG] switch_rtp.c:2323 Send middle packet
for [3] ts=341960 dur=480/480/800 seq=6961 lw=342280

2012-09-17 12:52:22.774919 [DEBUG] switch_rtp.c:2323 Send middle packet
for [3] ts=341960 dur=640/640/800 seq=6962 lw=342440

2012-09-17 12:52:22.794916 [DEBUG] switch_rtp.c:2323 Send end packet for
[3] ts=341960 dur=800/800/800 seq=6963 lw=342440

2012-09-17 12:52:22.794916 [DEBUG] switch_rtp.c:2323 Send end packet for
[3] ts=341960 dur=800/800/800 seq=6964 lw=342440

2012-09-17 12:52:22.794916 [DEBUG] switch_rtp.c:2323 Send end packet for
[3] ts=341960 dur=800/800/800 seq=6965 lw=342440

2012-09-17 12:52:22.794916 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:22.994918 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 5:800

2012-09-17 12:52:22.994918 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:23.334917 [DEBUG] switch_rtp.c:2420 Send start packet
for [4] ts=346920 dur=160/160/960 seq=6985 lw=346920

2012-09-17 12:52:23.354917 [DEBUG] switch_rtp.c:2323 Send middle packet
for [4] ts=346920 dur=320/320/960 seq=6986 lw=347080

2012-09-17 12:52:23.374915 [DEBUG] switch_rtp.c:2323 Send middle packet
for [4] ts=346920 dur=480/480/960 seq=6987 lw=347240

2012-09-17 12:52:23.394913 [DEBUG] switch_rtp.c:2323 Send middle packet
for [4] ts=346920 dur=640/640/960 seq=6988 lw=347400

2012-09-17 12:52:23.414914 [DEBUG] switch_rtp.c:2323 Send middle packet
for [4] ts=346920 dur=800/800/960 seq=6989 lw=347560

2012-09-17 12:52:23.434914 [DEBUG] switch_rtp.c:2323 Send end packet for
[4] ts=346920 dur=960/960/960 seq=6990 lw=347560

2012-09-17 12:52:23.434914 [DEBUG] switch_rtp.c:2323 Send end packet for
[4] ts=346920 dur=960/960/960 seq=6991 lw=347560

2012-09-17 12:52:23.434914 [DEBUG] switch_rtp.c:2323 Send end packet for
[4] ts=346920 dur=960/960/960 seq=6992 lw=347560

2012-09-17 12:52:23.434914 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:23.474915 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 6:800

2012-09-17 12:52:23.474915 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

nta: timer J fired, terminate 200 response

nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free

2012-09-17 12:52:23.974919 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 7:960

2012-09-17 12:52:23.974919 [DEBUG] switch_rtp.c:2420 Send start packet
for [5] ts=351560 dur=160/160/800 seq=7012 lw=351560

2012-09-17 12:52:23.974919 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:23.994918 [DEBUG] switch_rtp.c:2323 Send middle packet
for [5] ts=351560 dur=320/320/800 seq=7013 lw=351720

2012-09-17 12:52:24.014914 [DEBUG] switch_rtp.c:2323 Send middle packet
for [5] ts=351560 dur=480/480/800 seq=7014 lw=351880

2012-09-17 12:52:24.034914 [DEBUG] switch_rtp.c:2323 Send middle packet
for [5] ts=351560 dur=640/640/800 seq=7015 lw=352040

2012-09-17 12:52:24.054913 [DEBUG] switch_rtp.c:2323 Send end packet for
[5] ts=351560 dur=800/800/800 seq=7016 lw=352040

2012-09-17 12:52:24.054913 [DEBUG] switch_rtp.c:2323 Send end packet for
[5] ts=351560 dur=800/800/800 seq=7017 lw=352040

2012-09-17 12:52:24.054913 [DEBUG] switch_rtp.c:2323 Send end packet for
[5] ts=351560 dur=800/800/800 seq=7018 lw=352040

2012-09-17 12:52:24.054913 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:24.394917 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 8:960

2012-09-17 12:52:24.394917 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:24.594915 [DEBUG] switch_rtp.c:2420 Send start packet
for [6] ts=357000 dur=160/160/800 seq=7037 lw=357000

2012-09-17 12:52:24.614924 [DEBUG] switch_rtp.c:2323 Send middle packet
for [6] ts=357000 dur=320/320/800 seq=7038 lw=357160

2012-09-17 12:52:24.634922 [DEBUG] switch_rtp.c:2323 Send middle packet
for [6] ts=357000 dur=480/480/800 seq=7039 lw=357320

2012-09-17 12:52:24.654919 [DEBUG] switch_rtp.c:2323 Send middle packet
for [6] ts=357000 dur=640/640/800 seq=7040 lw=357480

2012-09-17 12:52:24.674918 [DEBUG] switch_rtp.c:2323 Send end packet for
[6] ts=357000 dur=800/800/800 seq=7041 lw=357480

2012-09-17 12:52:24.674918 [DEBUG] switch_rtp.c:2323 Send end packet for
[6] ts=357000 dur=800/800/800 seq=7042 lw=357480

2012-09-17 12:52:24.674918 [DEBUG] switch_rtp.c:2323 Send end packet for
[6] ts=357000 dur=800/800/800 seq=7043 lw=357480

2012-09-17 12:52:24.674918 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:24.814915 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 9:800

2012-09-17 12:52:24.814915 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:25.214926 [DEBUG] switch_rtp.c:2420 Send start packet
for [7] ts=361960 dur=160/160/960 seq=7063 lw=361960

2012-09-17 12:52:25.234921 [DEBUG] switch_rtp.c:2323 Send middle packet
for [7] ts=361960 dur=320/320/960 seq=7064 lw=362120

2012-09-17 12:52:25.254920 [DEBUG] switch_rtp.c:2323 Send middle packet
for [7] ts=361960 dur=480/480/960 seq=7065 lw=362280

2012-09-17 12:52:25.274919 [DEBUG] switch_rtp.c:2323 Send middle packet
for [7] ts=361960 dur=640/640/960 seq=7066 lw=362440

2012-09-17 12:52:25.294926 [DEBUG] switch_rtp.c:3410 RTP RECV DTMF 0:800

2012-09-17 12:52:25.294926 [DEBUG] switch_rtp.c:2323 Send middle packet
for [7] ts=361960 dur=800/800/960 seq=7067 lw=362600

2012-09-17 12:52:25.294926 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/internal/9910 at Kate-conf02-dev.webley.com:5080 [BREAK]

2012-09-17 12:52:25.314920 [DEBUG] switch_rtp.c:2323 Send end packet for
[7] ts=361960 dur=960/960/960 seq=7068 lw=362600

2012-09-17 12:52:25.314920 [DEBUG] switch_rtp.c:2323 Send end packet for
[7] ts=361960 dur=960/960/960 seq=7069 lw=362600

2012-09-17 12:52:25.314920 [DEBUG] switch_rtp.c:2323 Send end packet for
[7] ts=361960 dur=960/960/960 seq=7070 lw=362600

2012-09-17 12:52:25.314920 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:25.874915 [DEBUG] switch_rtp.c:2420 Send start packet
for [8] ts=161 dur=160/160/960 seq=7096 lw=161

2012-09-17 12:52:25.894918 [DEBUG] switch_rtp.c:2323 Send middle packet
for [8] ts=161 dur=320/320/960 seq=7097 lw=321

2012-09-17 12:52:25.914916 [DEBUG] switch_rtp.c:2323 Send middle packet
for [8] ts=161 dur=480/480/960 seq=7098 lw=481

2012-09-17 12:52:25.934917 [DEBUG] switch_rtp.c:2323 Send middle packet
for [8] ts=161 dur=640/640/960 seq=7099 lw=641

2012-09-17 12:52:25.954921 [DEBUG] switch_rtp.c:2323 Send middle packet
for [8] ts=161 dur=800/800/960 seq=7100 lw=801

2012-09-17 12:52:25.974930 [DEBUG] switch_rtp.c:2323 Send end packet for
[8] ts=161 dur=960/960/960 seq=7101 lw=801

2012-09-17 12:52:25.974930 [DEBUG] switch_rtp.c:2323 Send end packet for
[8] ts=161 dur=960/960/960 seq=7102 lw=801

2012-09-17 12:52:25.974930 [DEBUG] switch_rtp.c:2323 Send end packet for
[8] ts=161 dur=960/960/960 seq=7103 lw=801

2012-09-17 12:52:25.974930 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:26.514917 [DEBUG] switch_rtp.c:2420 Send start packet
for [9] ts=372360 dur=160/160/800 seq=7128 lw=372360

2012-09-17 12:52:26.534917 [DEBUG] switch_rtp.c:2323 Send middle packet
for [9] ts=372360 dur=320/320/800 seq=7129 lw=372520

2012-09-17 12:52:26.554924 [DEBUG] switch_rtp.c:2323 Send middle packet
for [9] ts=372360 dur=480/480/800 seq=7130 lw=372680

2012-09-17 12:52:26.574918 [DEBUG] switch_rtp.c:2323 Send middle packet
for [9] ts=372360 dur=640/640/800 seq=7131 lw=372840

2012-09-17 12:52:26.594915 [DEBUG] switch_rtp.c:2323 Send end packet for
[9] ts=372360 dur=800/800/800 seq=7132 lw=372840

2012-09-17 12:52:26.594915 [DEBUG] switch_rtp.c:2323 Send end packet for
[9] ts=372360 dur=800/800/800 seq=7133 lw=372840

2012-09-17 12:52:26.594915 [DEBUG] switch_rtp.c:2323 Send end packet for
[9] ts=372360 dur=800/800/800 seq=7134 lw=372840

2012-09-17 12:52:26.594915 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms

2012-09-17 12:52:27.134920 [DEBUG] switch_rtp.c:2420 Send start packet
for [0] ts=377480 dur=160/160/800 seq=7160 lw=377480

2012-09-17 12:52:27.154916 [DEBUG] switch_rtp.c:2323 Send middle packet
for [0] ts=377480 dur=320/320/800 seq=7161 lw=377640

2012-09-17 12:52:27.174916 [DEBUG] switch_rtp.c:2323 Send middle packet
for [0] ts=377480 dur=480/480/800 seq=7162 lw=377800

2012-09-17 12:52:27.194919 [DEBUG] switch_rtp.c:2323 Send middle packet
for [0] ts=377480 dur=640/640/800 seq=7163 lw=377960

2012-09-17 12:52:27.214916 [DEBUG] switch_rtp.c:2323 Send end packet for
[0] ts=377480 dur=800/800/800 seq=7164 lw=377960

2012-09-17 12:52:27.214916 [DEBUG] switch_rtp.c:2323 Send end packet for
[0] ts=377480 dur=800/800/800 seq=7165 lw=377960

2012-09-17 12:52:27.214916 [DEBUG] switch_rtp.c:2323 Send end packet for
[0] ts=377480 dur=800/800/800 seq=7166 lw=377960

2012-09-17 12:52:27.214916 [DEBUG] switch_rtp.c:2271 Queue digit delay
of 40ms


 
This message and any attachments to it are intended only for the addressee(s) identified above and may contain CONFIDENTIAL information. It is not intended for transmission to, or receipt by, any unauthorized persons.  If you are not an intended recipient or an agent responsible for delivering it to an intended recipient, you have received this e-mail in error and any dissemination, distribution, or copying of this message or any attachment to it is strictly prohibited. If you have received this email in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message from your system.
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From georg at riseup.net  Mon Sep 17 23:24:23 2012
From: georg at riseup.net (georg at riseup.net)
Date: Mon, 17 Sep 2012 21:24:23 +0200
Subject: [Freeswitch-users] snom call pickup blf
Message-ID: <f8a5df49516ccf20d487dcb7c961ed67.squirrel@fruiteater.riseup.net>

Hi,

Its been quite a long time, since you started the thread [1]. I've got the
same problem you've described. Did you solve this?

[1]
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/049500.html



From georg at riseup.net  Mon Sep 17 23:47:54 2012
From: georg at riseup.net (georg at riseup.net)
Date: Mon, 17 Sep 2012 21:47:54 +0200
Subject: [Freeswitch-users] snom call pickup blf
In-Reply-To: <f8a5df49516ccf20d487dcb7c961ed67.squirrel@fruiteater.riseup.net>
References: <f8a5df49516ccf20d487dcb7c961ed67.squirrel@fruiteater.riseup.net>
Message-ID: <36bae5d1f4b7e2e2b524dc07ef1cccb6.squirrel@fruiteater.riseup.net>

Sorry, this was not intended for the list...

> Hi,
>
> Its been quite a long time, since you started the thread [1]. I've got the
> same problem you've described. Did you solve this?
>
> [1]
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/049500.html
>






From paul at cupis.co.uk  Tue Sep 18 00:32:23 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 17 Sep 2012 21:32:23 +0100
Subject: [Freeswitch-users] SBC bridged call: DTMF delay
In-Reply-To: <74D0BA2985A4B04E8651FA01C8A58F3905E153FE@vs-win-ex01.corp.parusi.com>
References: <74D0BA2985A4B04E8651FA01C8A58F3905E153FE@vs-win-ex01.corp.parusi.com>
Message-ID: <505788D7.7060307@cupis.co.uk>

On 17/09/12 19:58, Alex Zarubin wrote:
> FS in SBC mode, ?external? and ?internal? sip calls are bridged. DTMF
> sequence ?11234567890? gets passed with a delay, - 2 sec between the
> last digit ?0? is received and sent (see log below). There is a
> consistent 500ms+ between ?Queue digit delay of 40ms? and ?Send start
> packet? for the next digit.
>
> Haven?t found a way to fix it so far (wiki, user list, parameters such
> as auto-rtp-bugs), help will be appreciated. CentOS 6.1, Xen, rfc2833.

Have you tried setting pass_rfc2833=true?

http://wiki.freeswitch.org/wiki/Variable_pass_rfc2833

Regards,


From ZAlex at webley.com  Tue Sep 18 01:12:22 2012
From: ZAlex at webley.com (Alex Zarubin)
Date: Mon, 17 Sep 2012 16:12:22 -0500
Subject: [Freeswitch-users] SBC bridged call: DTMF delay
References: <74D0BA2985A4B04E8651FA01C8A58F3905E153FE@vs-win-ex01.corp.parusi.com>
	<505788D7.7060307@cupis.co.uk>
Message-ID: <74D0BA2985A4B04E8651FA01C8A58F3905E15407@vs-win-ex01.corp.parusi.com>

Thank you Paul but it didn't work either. I'm still seeing inter-digit
500ms...

Alex

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
Cupis
Sent: Monday, September 17, 2012 3:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SBC bridged call: DTMF delay

On 17/09/12 19:58, Alex Zarubin wrote:
> FS in SBC mode, 'external' and 'internal' sip calls are bridged. DTMF 
> sequence '11234567890' gets passed with a delay, - 2 sec between the 
> last digit '0' is received and sent (see log below). There is a 
> consistent 500ms+ between 'Queue digit delay of 40ms' and 'Send start 
> packet' for the next digit.
>
> Haven't found a way to fix it so far (wiki, user list, parameters such

> as auto-rtp-bugs), help will be appreciated. CentOS 6.1, Xen, rfc2833.

Have you tried setting pass_rfc2833=true?

http://wiki.freeswitch.org/wiki/Variable_pass_rfc2833

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 and any attachments to it are intended only for the addressee(s) identified above and may contain CONFIDENTIAL information. It is not intended for transmission to, or receipt by, any unauthorized persons.  If you are not an intended recipient or an agent responsible for delivering it to an intended recipient, you have received this e-mail in error and any dissemination, distribution, or copying of this message or any attachment to it is strictly prohibited. If you have received this email in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message from your system.


From jmesquita at freeswitch.org  Tue Sep 18 02:42:41 2012
From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org)
Date: Mon, 17 Sep 2012 19:42:41 -0300
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <CAKzWOxWFgXz5fnDUStz4q53x0C3vRdrR4LFPsY2c9u6f69a2AA@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
	<CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>
	<CAKzWOxWFgXz5fnDUStz4q53x0C3vRdrR4LFPsY2c9u6f69a2AA@mail.gmail.com>
Message-ID: <17EE12D5-CC1D-49B4-829B-69EC21111547@freeswitch.org>

MC, I believe I have to make a statement here in favor of some realities that are different from the ones most people on this list live in. I live in South America and an IP Phone here does not get lower than 120usd per unit. And I am talking about the cheapest yea link phone model. In brazil, where I come from, this is even higher.

Taking that into consideration, several lower end ip phones started to appear as well as hybrid systems where the majority of the extensions are still analog. In a system like that, CTI applications like the one our friend is describing is really the way out to really add value to a solution.

Even on the asterisk world we see applications like these. For example the flash operator panel and the HUD for asterisk. I believe most of us have heard of it before.

Anyhow, I didn't mean to write an essay on the subject but I see this kind of feature being constantly rejected by the community and I really can't understand why that is.

Jo?o Mesquita

On 17/09/2012, at 01:39 p.m., Michael Collins <msc at freeswitch.org> wrote:

> Or get a hard phone that has a hold button and at least two line keys.
> -MC
> 
> On Mon, Sep 17, 2012 at 5:19 AM, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
> From personal experience, I believe that how you described is the right way to do it. The only caveat is that you will have to add variables do the channels so you can properly track what is going on in the cdrs . If you don't process cdrs, then it is all good. Careful with pickup and such too...
> 
> On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de> wrote:
> Hi MC,
> 
>  
> 
> thank you for the answer.
> 
> To your question, all call legs in this scenario (outgoing or incoming direction) are connected over a sip trunk of a pbx.
> 
>  
> 
> Scenario:
> 
> My CTI
> 
>   |  A
> 
>   |  |       event socket
> 
>   V  |
> 
> Freeswitch         Sip Trunk
> 
> <-                           PBX
> 
> ->                          
> 
>  
> 
> My Client control (CTI) the call legs over the event socket interface and the call legs are only legs over the sip trunk to or from the PBX.
> 
>  
> 
> After some tests i have found a possible solution:
> 
> -          Leg A and Leg B are bridged (all legs get the flag park_after_bridge = true)
> 
> -          For Consultation i park Leg B and transfer this Leg to Moh in my default context of my dialplan
> 
> -          i originate Leg C and bridge this Leg with Leg A (Leg C have the flag park_after_bridge = true too) Consultation is comlete now.
> 
> -          with the principle same think i can toggle Leg A <-> Leg B and Leg A <-> Leg C.
> 
>  
> 
> What is your think for this solution? I have tested this on the FS CLI and it works.
> 
>  
> 
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins
> Gesendet: Freitag, 14. September 2012 19:13
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Consultation Call via event_socket interface
> 
>  
> 
> Hi Alex,
> 
> Welcome to the FreeSWITCH mail list! 
> 
> First question for you: what kind of telephone are you using? The reason I ask is that this kind of function is trivially achieved with a good hard phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys. If you can use a hard phone w/ multiple line keys then you don't even need to mess with the dialplan, uuid_bridge, etc.
> 
> -MC
> 
> On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de> wrote:
> 
> Hi All,
> 
>  
> 
> I?m new on the mailing list.
> 
> I have a problem with a call scenario.
> 
> -          Channel A and channel B are bridged (A is my own channel and B is my calling partner)
> 
> -          Now i set channel B on hold with the command ?uuid_hold xxx? and create a new channel to C with the command:
> 
> bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip',origination_caller_id_number=num at ip}sofia/external/num at ip  &park
> 
> this works correctly, the partner C answer and the channel is established.
> 
> -          Now the Problem:
> 
> I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false.
> 
> Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment.
> 
> Is there a general problem in my plan to do that?
> 
> Is it a better plan to do this over the dialplan?
> 
> The next step in this scenario is to toggle the connection A -> B and A -> C.
> 
>  
> 
> Thanks for your help!
> 
> Nice regards,
> 
> 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
> 
> 
> 
> 
> -- 
> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Tue Sep 18 03:19:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Sep 2012 16:19:25 -0700
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <17EE12D5-CC1D-49B4-829B-69EC21111547@freeswitch.org>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
	<CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>
	<CAKzWOxWFgXz5fnDUStz4q53x0C3vRdrR4LFPsY2c9u6f69a2AA@mail.gmail.com>
	<17EE12D5-CC1D-49B4-829B-69EC21111547@freeswitch.org>
Message-ID: <CAKzWOxUEn9-46qQk2F3-=GonkUQrxfZThZ1-JDrSu-DiZjsicw@mail.gmail.com>

JM,

That actually explains a lot. I appreciate the information. I can
definitely see where the relative expense of a hard phone would make this
kind of application quite valuable. I hope you guys are able to iron out
the details. Also, maybe you could see if Mitch Capper could add line
appearances to FSClient. :)

-MC

On Mon, Sep 17, 2012 at 3:42 PM, Jmesquita at freeswitch.org <
jmesquita at freeswitch.org> wrote:

> MC, I believe I have to make a statement here in favor of some realities
> that are different from the ones most people on this list live in. I live
> in South America and an IP Phone here does not get lower than 120usd per
> unit. And I am talking about the cheapest yea link phone model. In brazil,
> where I come from, this is even higher.
>
> Taking that into consideration, several lower end ip phones started to
> appear as well as hybrid systems where the majority of the extensions are
> still analog. In a system like that, CTI applications like the one our
> friend is describing is really the way out to really add value to a
> solution.
>
> Even on the asterisk world we see applications like these. For example the
> flash operator panel and the HUD for asterisk. I believe most of us have
> heard of it before.
>
> Anyhow, I didn't mean to write an essay on the subject but I see this kind
> of feature being constantly rejected by the community and I really can't
> understand why that is.
>
> Jo?o Mesquita
>
> On 17/09/2012, at 01:39 p.m., Michael Collins <msc at freeswitch.org> wrote:
>
> Or get a hard phone that has a hold button and at least two line keys.
> -MC
>
> On Mon, Sep 17, 2012 at 5:19 AM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:
>
>> From personal experience, I believe that how you described is the right
>> way to do it. The only caveat is that you will have to add variables do the
>> channels so you can properly track what is going on in the cdrs . If you
>> don't process cdrs, then it is all good. Careful with pickup and such too...
>>  On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de>
>> wrote:
>>
>>>  Hi MC,****
>>>
>>> ** **
>>>
>>> thank you for the answer.****
>>>
>>> To your question, all call legs in this scenario (outgoing or incoming
>>> direction) are connected over a sip trunk of a pbx.****
>>>
>>> ** **
>>>
>>> Scenario:****
>>>
>>> My CTI****
>>>
>>>   |  A****
>>>
>>>   |  |       event socket****
>>>
>>>   V  |****
>>>
>>> Freeswitch         Sip Trunk****
>>>
>>> <-                           PBX****
>>>
>>> ->                           ****
>>>
>>> ** **
>>>
>>> My Client control (CTI) the call legs over the event socket interface
>>> and the call legs are only legs over the sip trunk to or from the PBX.**
>>> **
>>>
>>> ** **
>>>
>>> After some tests i have found a possible solution:****
>>>
>>> **-          **Leg A and Leg B are bridged (all legs get the flag
>>> park_after_bridge = true)****
>>>
>>> **-          **For Consultation i park Leg B and transfer this Leg to
>>> Moh in my default context of my dialplan****
>>>
>>> **-          **i originate Leg C and bridge this Leg with Leg A (Leg C
>>> have the flag park_after_bridge = true too) Consultation is comlete now.
>>> ****
>>>
>>> **-          **with the principle same think i can toggle Leg A <-> Leg
>>> B and Leg A <-> Leg C.****
>>>
>>> ** **
>>>
>>> What is your think for this solution? I have tested this on the FS CLI
>>> and it works.****
>>>
>>> ** **
>>>
>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Michael
>>> Collins
>>> *Gesendet:* Freitag, 14. September 2012 19:13
>>> *An:* FreeSWITCH Users Help
>>> *Betreff:* Re: [Freeswitch-users] Consultation Call via event_socket
>>> interface****
>>>
>>> ** **
>>>
>>> Hi Alex,
>>>
>>> Welcome to the FreeSWITCH mail list!
>>>
>>> First question for you: what kind of telephone are you using? The reason
>>> I ask is that this kind of function is trivially achieved with a good hard
>>> phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys.
>>> If you can use a hard phone w/ multiple line keys then you don't even need
>>> to mess with the dialplan, uuid_bridge, etc.
>>>
>>> -MC****
>>>
>>> On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <
>>> Alexander.Haugg at c4b.de> wrote:****
>>>
>>> Hi All,****
>>>
>>>  ****
>>>
>>> I?m new on the mailing list.****
>>>
>>> I have a problem with a call scenario.****
>>>
>>> -          Channel A and channel B are bridged (A is my own channel and
>>> B is my calling partner)****
>>>
>>> -          Now i set channel B on hold with the command ?uuid_hold xxx?
>>> and create a new channel to C with the command:****
>>>
>>> bgapi originate
>>> {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip
>>> ',origination_caller_id_number=num at ip}sofia/external/num at ip  &park****
>>>
>>> this works correctly, the partner C answer and the channel is
>>> established.****
>>>
>>> -          Now the Problem:****
>>>
>>> I try to bridge the channel a to channel c with the uuid_bridge command,
>>> now the channel b will hangup, why? The variable hangup_after_bridge is by
>>> default false.****
>>>
>>> Other problem: channel A can hear the voice of channel C but not speak
>>> with him, channel C can hear and speak. But this problem is not the
>>> important think at the moment.****
>>>
>>> Is there a general problem in my plan to do that?****
>>>
>>> Is it a better plan to do this over the dialplan?****
>>>
>>> The next step in this scenario is to toggle the connection A -> B and A
>>> -> C.****
>>>
>>>  ****
>>>
>>> Thanks for your help!****
>>>
>>> Nice regards,****
>>>
>>> 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****
>>>
>>>
>>>
>>>
>>> --
>>> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
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> http://www.freeswitch.org
>
>
> _________________________________________________________________________
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> 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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> 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 dujinfang at gmail.com  Tue Sep 18 03:49:57 2012
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 18 Sep 2012 07:49:57 +0800
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <CAKzWOxUEn9-46qQk2F3-=GonkUQrxfZThZ1-JDrSu-DiZjsicw@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
	<CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>
	<CAKzWOxWFgXz5fnDUStz4q53x0C3vRdrR4LFPsY2c9u6f69a2AA@mail.gmail.com>
	<17EE12D5-CC1D-49B4-829B-69EC21111547@freeswitch.org>
	<CAKzWOxUEn9-46qQk2F3-=GonkUQrxfZThZ1-JDrSu-DiZjsicw@mail.gmail.com>
Message-ID: <20E48A825239435BB030CC5C30565542@gmail.com>

I had the similar issue, when using mod_fifo with mixed inbound/outbound agents, I have to disable the multiline feature on hard phones (any one knows how to disable on eye beam or xlite?)

att_xfer could do this job except it has problems when use with event_socket http://jira.freeswitch.org/browse/FS-4419

, maybe it's not designed to this kind of usage.

Yes, I would use the similar approach with Alexander if I'm doing this. Also I wonder if it's possible to just use a conference with mute/unmute hear/nohear and relate/unrelate combinations.


--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, September 18, 2012 at 7:19 AM, Michael Collins wrote:

> JM,
>  
> That actually explains a lot. I appreciate the information. I can definitely see where the relative expense of a hard phone would make this kind of application quite valuable. I hope you guys are able to iron out the details. Also, maybe you could see if Mitch Capper could add line appearances to FSClient. :)
>  
> -MC
>  
> On Mon, Sep 17, 2012 at 3:42 PM, Jmesquita at freeswitch.org (mailto:Jmesquita at freeswitch.org) <jmesquita at freeswitch.org (mailto:jmesquita at freeswitch.org)> wrote:
> > MC, I believe I have to make a statement here in favor of some realities that are different from the ones most people on this list live in. I live in South America and an IP Phone here does not get lower than 120usd per unit. And I am talking about the cheapest yea link phone model. In brazil, where I come from, this is even higher.  
> >  
> > Taking that into consideration, several lower end ip phones started to appear as well as hybrid systems where the majority of the extensions are still analog. In a system like that, CTI applications like the one our friend is describing is really the way out to really add value to a solution.  
> >  
> > Even on the asterisk world we see applications like these. For example the flash operator panel and the HUD for asterisk. I believe most of us have heard of it before.
> >  
> > Anyhow, I didn't mean to write an essay on the subject but I see this kind of feature being constantly rejected by the community and I really can't understand why that is.  
> >  
> > Jo?o Mesquita
> >  
> > On 17/09/2012, at 01:39 p.m., Michael Collins <msc at freeswitch.org (mailto:msc at freeswitch.org)> wrote:
> >  
> > > Or get a hard phone that has a hold button and at least two line keys.
> > > -MC
> > >  
> > > On Mon, Sep 17, 2012 at 5:19 AM, Jo?o Mesquita <jmesquita at freeswitch.org (mailto:jmesquita at freeswitch.org)> wrote:
> > > > From personal experience, I believe that how you described is the right way to do it. The only caveat is that you will have to add variables do the channels so you can properly track what is going on in the cdrs . If you don't process cdrs, then it is all good. Careful with pickup and such too...  
> > > > On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de (mailto:Alexander.Haugg at c4b.de)> wrote:
> > > > > Hi MC,  
> > > > >   
> > > > > thank you for the answer.
> > > > > To your question, all call legs in this scenario (outgoing or incoming direction) are connected over a sip trunk of a pbx.
> > > > >   
> > > > > Scenario:
> > > > > My CTI
> > > > >   |  A
> > > > >   |  |       event socket
> > > > >   V  |
> > > > > Freeswitch         Sip Trunk
> > > > > <-                           PBX
> > > > > ->                            
> > > > >   
> > > > > My Client control (CTI) the call legs over the event socket interface and the call legs are only legs over the sip trunk to or from the PBX.
> > > > >   
> > > > > After some tests i have found a possible solution:
> > > > > -          Leg A and Leg B are bridged (all legs get the flag park_after_bridge = true)
> > > > > -          For Consultation i park Leg B and transfer this Leg to Moh in my default context of my dialplan
> > > > > -          i originate Leg C and bridge this Leg with Leg A (Leg C have the flag park_after_bridge = true too) Consultation is comlete now.
> > > > > -          with the principle same think i can toggle Leg A <-> Leg B and Leg A <-> Leg C.
> > > > >   
> > > > > What is your think for this solution? I have tested this on the FS CLI and it works.
> > > > >   
> > > > > Von: 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 Michael Collins
> > > > > Gesendet: Freitag, 14. September 2012 19:13
> > > > > An: FreeSWITCH Users Help
> > > > > Betreff: Re: [Freeswitch-users] Consultation Call via event_socket interface  
> > > > >   
> > > > > Hi Alex,
> > > > >  
> > > > > Welcome to the FreeSWITCH mail list!  
> > > > >  
> > > > > First question for you: what kind of telephone are you using? The reason I ask is that this kind of function is trivially achieved with a good hard phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys. If you can use a hard phone w/ multiple line keys then you don't even need to mess with the dialplan, uuid_bridge, etc.
> > > > >  
> > > > > -MC  
> > > > > On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de (mailto:Alexander.Haugg at c4b.de)> wrote:
> > > > > Hi All,
> > > > >   
> > > > > I?m new on the mailing list.
> > > > > I have a problem with a call scenario.
> > > > > -          Channel A and channel B are bridged (A is my own channel and B is my calling partner)
> > > > > -          Now i set channel B on hold with the command ?uuid_hold xxx? and create a new channel to C with the command:
> > > > > bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip',origination_caller_id_number=num at ip}sofia/external/num at ip  &park
> > > > > this works correctly, the partner C answer and the channel is established.
> > > > > -          Now the Problem:
> > > > > I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false.
> > > > > Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment.
> > > > > Is there a general problem in my plan to do that?
> > > > > Is it a better plan to do this over the dialplan?
> > > > > The next step in this scenario is to toggle the connection A -> B and A -> C.
> > > > >   
> > > > > Thanks for your help!
> > > > > Nice regards,
> > > > > Alex
> > > > >  
> > > > >  
> > > > >  
> > > > > _________________________________________________________________________
> > > > > 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 (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 (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
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>  
>  


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From anthony.minessale at gmail.com  Tue Sep 18 05:02:05 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Sep 2012 20:02:05 -0500
Subject: [Freeswitch-users] t.38 re-invite results in 407 Proxy
 Authentication Required
In-Reply-To: <5057418A.6010006@xpirio.com>
References: <5057418A.6010006@xpirio.com>
Message-ID: <CAKbxfG_J0bfLBU+E3Qt3UkcD=07nkQhBcFiVf_D_PkUyi98R9w@mail.gmail.com>

This is a pretty big issue, fixed in latest 1.3

On Mon, Sep 17, 2012 at 10:28 AM, Christian L?schenkohl <
christian.loeschenkohl at xpirio.com> wrote:

> hello fs users
>
> anybody have similar problems than me?
>
> symptom is that within a t.38 re-invite freeswitch responds with 407 Proxy
> Authentication Required
> wasn't the case in e.g. 1.2.0-rc2
> now with 1.2.2+git~20120911T185917Z~3fd1a57902 it's like described
>
> was this an intended changed. if yes - why?
> many sip ua devices only respond with an ACK instead of an re-invite with
> user credentials.
>
> br
>
> --
> Ing. Christian L?schenkohl
> Senior VoIP Engineer
> Head of VoIP Engineering
> Head of R&D
>
> T +43 5 77 11 - 1000
> F +43 5 77 11 - 1002
> E christian.loeschenkohl at xpirio.com
> W www.xpirio.com
>
> xpirio Telekommunikation & Service GmbH
>
> Gerichtsstand Klagenfurt FN297465z
> Lakeside B04 | 9020 Klagenfurt | Austria
>
> An ESS Group company
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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From jaganthoutam at gmail.com  Tue Sep 18 06:27:10 2012
From: jaganthoutam at gmail.com (Jagadish Thoutam)
Date: Mon, 17 Sep 2012 19:27:10 -0700
Subject: [Freeswitch-users] issue with web server to handle XML CDRs
In-Reply-To: <CAKzWOxXz5aCKXBFt0a1wFysMUC6_wCzJCC=M9usJqKHPKfgXRw@mail.gmail.com>
References: <CAHqn9HFfo9JiB5mz747Pg18q5hPZXgfV6cnroyQF_SqBkcepzA@mail.gmail.com>
	<CAKzWOxXz5aCKXBFt0a1wFysMUC6_wCzJCC=M9usJqKHPKfgXRw@mail.gmail.com>
Message-ID: <CAHqn9HF7iZtgdBkPDxdP3QSTAeEEMmqvxUAiocpP_4q1-8=CMA@mail.gmail.com>

Hi Michael,

   There are no errors in fs_cli i thing  the request is not reaching to
web server.

On 17 September 2012 09:37, Michael Collins <msc at freeswitch.org> wrote:

> Next step  is to watch the fs_cli to see if there are any errors. Also,
> look at your web server access and error logs to see if the request is
> making it there.
> -MC
>
>  On Mon, Sep 17, 2012 at 2:36 AM, Jagadish Thoutam <jaganthoutam at gmail.com
> > wrote:
>
>>  Hi all,
>>
>>    There is the issue with xml_cdr for me when i try to call <param
>> name="url" value="http://localhost/cgi-bin/cdr.pl"/> its not working, i
>> have load the mod_xml_cdr on cli, and my cdr.pl file is working well
>> when try it manually and also have executable permission for cdr.plfile, please some one tell me what i am missing here.
>>
>>
>>
>> Thanks
>>
>> Jagadish
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <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 lists at kavun.ch  Tue Sep 18 06:37:01 2012
From: lists at kavun.ch (Emrah)
Date: Mon, 17 Sep 2012 22:37:01 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <CAKzWOxX0ck=HWZzQp7sg0xbL9AZYRrrfJNuSKYbW0XMqHMoNHw@mail.gmail.com>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
	<ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
	<002a01cd94d5$3c0a3820$b41ea860$@gmail.com>
	<0A200C25-FFE2-4D2F-84C8-7DD3B53007EB@kavun.ch>
	<CAKzWOxX0ck=HWZzQp7sg0xbL9AZYRrrfJNuSKYbW0XMqHMoNHw@mail.gmail.com>
Message-ID: <28E03C16-8034-4034-8983-B9809D061F68@kavun.ch>

Hey Michael, thanks a bunch for testing.
Jira is not screenreader friendly and I am having a bit of trouble creating the issue. 

I would be super grateful if someone could create it for me.

Best,
E
On Sep 17, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:

> FTR, I was able to reproduce this behavior. If you haven't already done so you can file a jira.
> -MC
> 
> On Mon, Sep 17, 2012 at 7:53 AM, Emrah <lists at kavun.ch> wrote:
> Hey there,
> 
> I use RFC2833 directly from my hardware phone to FS. The issue is not client dependent and happens with any UA.
> Here is the dialplan sequence that triggers to the play_and_get_digits
> 
> <action application="answer" />
> <action application="sleep" data="1000" />
> <action application="play_and_get_digits" data="4 5 3 10000 # /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/pls-enter-conf-room-number.wav /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/conf-number-invalid-contact-conf-administrator.wav conf_num ^30\d{2}$ 3000" />
> 
> Any idea would be greatly appreciated.
> 
> Best,
> Emrah
> On Sep 17, 2012, at 9:06 AM, grmt <garmt.noname at gmail.com> wrote:
> 
> > I didn't spend time (yet) to reproduce your issue, but you may want to give
> > us some more information:
> >
> > What kind of DTMF (inband /outband - RFC2833) are you using, what is your
> > client?
> > Are you testing locally or through a provider?
> > Maybe share your dialplan/script ...
> > Are you using text-to-speech?
> >
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gabe at gundy.org  Tue Sep 18 09:08:22 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 17 Sep 2012 23:08:22 -0600
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: <CAMwZ-tKX1Pdn2vgqnXghkpXSrK-7yEQw4qbZZ6mwJUOFkxAcxg@mail.gmail.com>

On Tue, Aug 28, 2012 at 9:28 AM, Babak Yakhchali
<babak.freeswitch at gmail.com> wrote:
> 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.

What about this?

bridge(
[execute_on_originate=set_user::SOMEUSER at DOMAIN]  user/SOMEUSER at DOMAIN
)

...or something.

Best,
Gabe


From anton.jugatsu at gmail.com  Tue Sep 18 09:49:05 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 18 Sep 2012 09:49:05 +0400
Subject: [Freeswitch-users] t.38 re-invite results in 407 Proxy
 Authentication Required
In-Reply-To: <CAKbxfG_J0bfLBU+E3Qt3UkcD=07nkQhBcFiVf_D_PkUyi98R9w@mail.gmail.com>
References: <5057418A.6010006@xpirio.com>
	<CAKbxfG_J0bfLBU+E3Qt3UkcD=07nkQhBcFiVf_D_PkUyi98R9w@mail.gmail.com>
Message-ID: <CANBhyPC7mH8Zb29vFhHUfLzww3eCRjeEt3OOgwr=cgThdepMEg@mail.gmail.com>

Would it be backported to stable branch?

2012/9/18 Anthony Minessale <anthony.minessale at gmail.com>

> This is a pretty big issue, fixed in latest 1.3
>
>
> On Mon, Sep 17, 2012 at 10:28 AM, Christian L?schenkohl <
> christian.loeschenkohl at xpirio.com> wrote:
>
>> hello fs users
>>
>> anybody have similar problems than me?
>>
>> symptom is that within a t.38 re-invite freeswitch responds with 407
>> Proxy Authentication Required
>> wasn't the case in e.g. 1.2.0-rc2
>> now with 1.2.2+git~20120911T185917Z~3fd1a57902 it's like described
>>
>> was this an intended changed. if yes - why?
>> many sip ua devices only respond with an ACK instead of an re-invite with
>> user credentials.
>>
>> br
>>
>> --
>> Ing. Christian L?schenkohl
>> Senior VoIP Engineer
>> Head of VoIP Engineering
>> Head of R&D
>>
>> T +43 5 77 11 - 1000
>> F +43 5 77 11 - 1002
>> E christian.loeschenkohl at xpirio.com
>> W www.xpirio.com
>>
>> xpirio Telekommunikation & Service GmbH
>>
>> Gerichtsstand Klagenfurt FN297465z
>> Lakeside B04 | 9020 Klagenfurt | Austria
>>
>> An ESS Group company
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
-------------- next part --------------
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From covici at ccs.covici.com  Tue Sep 18 10:17:15 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 18 Sep 2012 02:17:15 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <28E03C16-8034-4034-8983-B9809D061F68@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
	<ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
	<002a01cd94d5$3c0a3820$b41ea860$@gmail.com>
	<0A200C25-FFE2-4D2F-84C8-7DD3B53007EB@kavun.ch>
	<CAKzWOxX0ck=HWZzQp7sg0xbL9AZYRrrfJNuSKYbW0XMqHMoNHw@mail.gmail.com>
	<28E03C16-8034-4034-8983-B9809D061F68@kavun.ch>
Message-ID: <965.1347949035@ccs.covici.com>

What screen reader are you using -- its definitely a pain, but can be
done.

Emrah <lists at kavun.ch> wrote:

> Hey Michael, thanks a bunch for testing.
> Jira is not screenreader friendly and I am having a bit of trouble creating the issue. 
> 
> I would be super grateful if someone could create it for me.
> 
> Best,
> E
> On Sep 17, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
> > FTR, I was able to reproduce this behavior. If you haven't already done so you can file a jira.
> > -MC
> > 
> > On Mon, Sep 17, 2012 at 7:53 AM, Emrah <lists at kavun.ch> wrote:
> > Hey there,
> > 
> > I use RFC2833 directly from my hardware phone to FS. The issue is not client dependent and happens with any UA.
> > Here is the dialplan sequence that triggers to the play_and_get_digits
> > 
> > <action application="answer" />
> > <action application="sleep" data="1000" />
> > <action application="play_and_get_digits" data="4 5 3 10000 # /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/pls-enter-conf-room-number.wav /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/conf-number-invalid-contact-conf-administrator.wav conf_num ^30\d{2}$ 3000" />
> > 
> > Any idea would be greatly appreciated.
> > 
> > Best,
> > Emrah
> > On Sep 17, 2012, at 9:06 AM, grmt <garmt.noname at gmail.com> wrote:
> > 
> > > I didn't spend time (yet) to reproduce your issue, but you may want to give
> > > us some more information:
> > >
> > > What kind of DTMF (inband /outband - RFC2833) are you using, what is your
> > > client?
> > > Are you testing locally or through a provider?
> > > Maybe share your dialplan/script ...
> > > Are you using text-to-speech?
> > >
> > >
> > >
> > >
> > >
> > >
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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

-- 
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 miha at softnet.si  Tue Sep 18 13:09:36 2012
From: miha at softnet.si (Miha)
Date: Tue, 18 Sep 2012 11:09:36 +0200
Subject: [Freeswitch-users] xml curl
Message-ID: <50583A50.8020405@softnet.si>

hi,

is it possible to use xml_curl just for a one part of my dialplan?

Regards,
Miha


From lists at kavun.ch  Tue Sep 18 14:58:05 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 18 Sep 2012 06:58:05 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <965.1347949035@ccs.covici.com>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
	<ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
	<002a01cd94d5$3c0a3820$b41ea860$@gmail.com>
	<0A200C25-FFE2-4D2F-84C8-7DD3B53007EB@kavun.ch>
	<CAKzWOxX0ck=HWZzQp7sg0xbL9AZYRrrfJNuSKYbW0XMqHMoNHw@mail.gmail.com>
	<28E03C16-8034-4034-8983-B9809D061F68@kavun.ch>
	<965.1347949035@ccs.covici.com>
Message-ID: <3F23ACEB-8A0E-4E46-B0B0-227146812E75@kavun.ch>

I'm using VO on a Mac. It's mainly the input fields / optional lists that are annoying. E.g.: the component field. Using down arrow doesn't show me any suggestion.
On Sep 18, 2012, at 2:17 AM, covici at ccs.covici.com wrote:

> What screen reader are you using -- its definitely a pain, but can be
> done.
> 
> Emrah <lists at kavun.ch> wrote:
> 
>> Hey Michael, thanks a bunch for testing.
>> Jira is not screenreader friendly and I am having a bit of trouble creating the issue. 
>> 
>> I would be super grateful if someone could create it for me.
>> 
>> Best,
>> E
>> On Sep 17, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> FTR, I was able to reproduce this behavior. If you haven't already done so you can file a jira.
>>> -MC
>>> 
>>> On Mon, Sep 17, 2012 at 7:53 AM, Emrah <lists at kavun.ch> wrote:
>>> Hey there,
>>> 
>>> I use RFC2833 directly from my hardware phone to FS. The issue is not client dependent and happens with any UA.
>>> Here is the dialplan sequence that triggers to the play_and_get_digits
>>> 
>>> <action application="answer" />
>>> <action application="sleep" data="1000" />
>>> <action application="play_and_get_digits" data="4 5 3 10000 # /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/pls-enter-conf-room-number.wav /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/conf-number-invalid-contact-conf-administrator.wav conf_num ^30\d{2}$ 3000" />
>>> 
>>> Any idea would be greatly appreciated.
>>> 
>>> Best,
>>> Emrah
>>> On Sep 17, 2012, at 9:06 AM, grmt <garmt.noname at gmail.com> wrote:
>>> 
>>>> I didn't spend time (yet) to reproduce your issue, but you may want to give
>>>> us some more information:
>>>> 
>>>> What kind of DTMF (inband /outband - RFC2833) are you using, what is your
>>>> client?
>>>> Are you testing locally or through a provider?
>>>> Maybe share your dialplan/script ...
>>>> Are you using text-to-speech?
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
> 
> -- 
> 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



From kbdfck at gmail.com  Tue Sep 18 15:24:23 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 18 Sep 2012 15:24:23 +0400
Subject: [Freeswitch-users] t.38 re-invite results in 407 Proxy
 Authentication Required
In-Reply-To: <CANBhyPC7mH8Zb29vFhHUfLzww3eCRjeEt3OOgwr=cgThdepMEg@mail.gmail.com>
References: <5057418A.6010006@xpirio.com>
	<CAKbxfG_J0bfLBU+E3Qt3UkcD=07nkQhBcFiVf_D_PkUyi98R9w@mail.gmail.com>
	<CANBhyPC7mH8Zb29vFhHUfLzww3eCRjeEt3OOgwr=cgThdepMEg@mail.gmail.com>
Message-ID: <CA+GmxbYAL8V7PxAFoJCL6XxVK2KgoxLG2yeE9-NdjZk856o5Sg@mail.gmail.com>

We have same issue, will try update to latest today

2012/9/18 Anton Kvashenkin <anton.jugatsu at gmail.com>

> Would it be backported to stable branch?
>
>
> 2012/9/18 Anthony Minessale <anthony.minessale at gmail.com>
>
>> This is a pretty big issue, fixed in latest 1.3
>>
>>
>> On Mon, Sep 17, 2012 at 10:28 AM, Christian L?schenkohl <
>> christian.loeschenkohl at xpirio.com> wrote:
>>
>>> hello fs users
>>>
>>> anybody have similar problems than me?
>>>
>>> symptom is that within a t.38 re-invite freeswitch responds with 407
>>> Proxy Authentication Required
>>> wasn't the case in e.g. 1.2.0-rc2
>>> now with 1.2.2+git~20120911T185917Z~3fd1a57902 it's like described
>>>
>>> was this an intended changed. if yes - why?
>>> many sip ua devices only respond with an ACK instead of an re-invite
>>> with user credentials.
>>>
>>> br
>>>
>>> --
>>> Ing. Christian L?schenkohl
>>> Senior VoIP Engineer
>>> Head of VoIP Engineering
>>> Head of R&D
>>>
>>> T +43 5 77 11 - 1000
>>> F +43 5 77 11 - 1002
>>> E christian.loeschenkohl at xpirio.com
>>> W www.xpirio.com
>>>
>>> xpirio Telekommunikation & Service GmbH
>>>
>>> Gerichtsstand Klagenfurt FN297465z
>>> Lakeside B04 | 9020 Klagenfurt | Austria
>>>
>>> An ESS Group company
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From covici at ccs.covici.com  Tue Sep 18 16:05:09 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 18 Sep 2012 08:05:09 -0400
Subject: [Freeswitch-users] Issue with play_and_get_digits
In-Reply-To: <3F23ACEB-8A0E-4E46-B0B0-227146812E75@kavun.ch>
References: <927E87DB-16F3-4865-A231-85F0A89FCC14@kavun.ch>
	<CAKzWOxVs4y9DSRnwm4z7fhzPEnP_SYh-sv3k0j9qkTNhO0=dYw@mail.gmail.com>
	<34980B88-DB29-4CF4-8A79-44F340B65AF7@kavun.ch>
	<7CBD3D5E-D7BE-447D-A605-E371F2353962@kavun.ch>
	<27905.1347525413@ccs.covici.com>
	<A436ADC4-2C1F-460A-A2C7-873F6A3A2035@kavun.ch>
	<ACB88664-BB9E-45E6-81FC-87E27A1A50DA@kavun.ch>
	<002a01cd94d5$3c0a3820$b41ea860$@gmail.com>
	<0A200C25-FFE2-4D2F-84C8-7DD3B53007EB@kavun.ch>
	<CAKzWOxX0ck=HWZzQp7sg0xbL9AZYRrrfJNuSKYbW0XMqHMoNHw@mail.gmail.com>
	<28E03C16-8034-4034-8983-B9809D061F68@kavun.ch>
	<965.1347949035@ccs.covici.com>
	<3F23ACEB-8A0E-4E46-B0B0-227146812E75@kavun.ch>
Message-ID: <14817.1347969909@ccs.covici.com>

Does the list say popup or anything?  I am using WindowEyes and I was
able to navigate the list that you mention successfully, but I have
never tried on a MAC.  You might want to make sure you are on the latest
version of the OS, they have made some small improvements along the way.

Emrah <lists at kavun.ch> wrote:

> I'm using VO on a Mac. It's mainly the input fields / optional lists that are annoying. E.g.: the component field. Using down arrow doesn't show me any suggestion.
> On Sep 18, 2012, at 2:17 AM, covici at ccs.covici.com wrote:
> 
> > What screen reader are you using -- its definitely a pain, but can be
> > done.
> > 
> > Emrah <lists at kavun.ch> wrote:
> > 
> >> Hey Michael, thanks a bunch for testing.
> >> Jira is not screenreader friendly and I am having a bit of trouble creating the issue. 
> >> 
> >> I would be super grateful if someone could create it for me.
> >> 
> >> Best,
> >> E
> >> On Sep 17, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
> >> 
> >>> FTR, I was able to reproduce this behavior. If you haven't already done so you can file a jira.
> >>> -MC
> >>> 
> >>> On Mon, Sep 17, 2012 at 7:53 AM, Emrah <lists at kavun.ch> wrote:
> >>> Hey there,
> >>> 
> >>> I use RFC2833 directly from my hardware phone to FS. The issue is not client dependent and happens with any UA.
> >>> Here is the dialplan sequence that triggers to the play_and_get_digits
> >>> 
> >>> <action application="answer" />
> >>> <action application="sleep" data="1000" />
> >>> <action application="play_and_get_digits" data="4 5 3 10000 # /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/pls-enter-conf-room-number.wav /usr/local/freeswitch/sounds/en/us/emrah/custom/8000/conf-number-invalid-contact-conf-administrator.wav conf_num ^30\d{2}$ 3000" />
> >>> 
> >>> Any idea would be greatly appreciated.
> >>> 
> >>> Best,
> >>> Emrah
> >>> On Sep 17, 2012, at 9:06 AM, grmt <garmt.noname at gmail.com> wrote:
> >>> 
> >>>> I didn't spend time (yet) to reproduce your issue, but you may want to give
> >>>> us some more information:
> >>>> 
> >>>> What kind of DTMF (inband /outband - RFC2833) are you using, what is your
> >>>> client?
> >>>> Are you testing locally or through a provider?
> >>>> Maybe share your dialplan/script ...
> >>>> Are you using text-to-speech?
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>> 
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>> 
> >>> 
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>> 
> >>> 
> >>> 
> >>> 
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>> 
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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
> > 
> > -- 
> > 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

-- 
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  Tue Sep 18 17:46:54 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 18 Sep 2012 08:46:54 -0500
Subject: [Freeswitch-users] t.38 re-invite results in 407 Proxy
	Authentication Required
In-Reply-To: <CA+GmxbYAL8V7PxAFoJCL6XxVK2KgoxLG2yeE9-NdjZk856o5Sg@mail.gmail.com>
References: <5057418A.6010006@xpirio.com>
	<CAKbxfG_J0bfLBU+E3Qt3UkcD=07nkQhBcFiVf_D_PkUyi98R9w@mail.gmail.com>
	<CANBhyPC7mH8Zb29vFhHUfLzww3eCRjeEt3OOgwr=cgThdepMEg@mail.gmail.com>
	<CA+GmxbYAL8V7PxAFoJCL6XxVK2KgoxLG2yeE9-NdjZk856o5Sg@mail.gmail.com>
Message-ID: <1C9FBB6F-95EC-41B2-A884-9A2F6D1288AB@freeswitch.org>

yes this fix is being backported... should be there today with a pile of other thingw

Ken
Sent from my iPad

On Sep 18, 2012, at 6:24 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> We have same issue, will try update to latest today
> 
> 2012/9/18 Anton Kvashenkin <anton.jugatsu at gmail.com>
> Would it be backported to stable branch?
> 
> 
> 2012/9/18 Anthony Minessale <anthony.minessale at gmail.com>
> This is a pretty big issue, fixed in latest 1.3 
> 
> 
> On Mon, Sep 17, 2012 at 10:28 AM, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:
> hello fs users
> 
> anybody have similar problems than me?
> 
> symptom is that within a t.38 re-invite freeswitch responds with 407 Proxy Authentication Required
> wasn't the case in e.g. 1.2.0-rc2
> now with 1.2.2+git~20120911T185917Z~3fd1a57902 it's like described
> 
> was this an intended changed. if yes - why?
> many sip ua devices only respond with an ACK instead of an re-invite with user credentials.
> 
> br
> 
> --
> Ing. Christian L?schenkohl
> Senior VoIP Engineer
> Head of VoIP Engineering
> Head of R&D
> 
> T +43 5 77 11 - 1000
> F +43 5 77 11 - 1002
> E christian.loeschenkohl at xpirio.com
> W www.xpirio.com
> 
> xpirio Telekommunikation & Service GmbH
> 
> Gerichtsstand Klagenfurt FN297465z
> Lakeside B04 | 9020 Klagenfurt | Austria
> 
> An ESS Group company
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
> 
> 
> 
> 
> -- 
> Best regards,
> 
> Dmitry Sytchev,
> IT Engineer
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bigx333 at gmail.com  Tue Sep 18 14:33:14 2012
From: bigx333 at gmail.com (Nelson Camargo)
Date: Tue, 18 Sep 2012 12:33:14 +0200
Subject: [Freeswitch-users] xml curl
In-Reply-To: <50583A50.8020405@softnet.si>
References: <50583A50.8020405@softnet.si>
Message-ID: <5AE42E9E-D786-442D-A41F-3414F194025C@gmail.com>

Yes, if the freeswitch can't find a route in the xml response it will fall back to the .xml file.

On 18 Sep 2012, at 11:09 AM, Miha wrote:

> hi,
> 
> is it possible to use xml_curl just for a one part of my dialplan?
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From peetzer at gmail.com  Tue Sep 18 14:57:54 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Tue, 18 Sep 2012 12:57:54 +0200
Subject: [Freeswitch-users] Mod xml curl and mod dingaling
Message-ID: <CACSMF4wUjNMat+N6f3MpUzJty=hgpOzoV2Bg2Sp069FcX1w87A@mail.gmail.com>

Hi all,

I'm setting up FreeSwitch and I'm using mod xml curl.

I catch all calls and redirect them (post from Freeswitch to my servlet) to
my handler.

I give all 'sip' extension configurations dynamically back in xml.

How does the complete configuration look of a basic Gtalk client?

http://wiki.freeswitch.org/wiki/Dingaling#Sample_Configuration_.28Google.29

Can you include the extension in the client profile?

Kind regards,

Peter

Ps, is it also possible to stop automatic posts to my handler? I want to be
able to control this manually by runing for example "reloadacl" command.
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From oliver.saggau at sysvision.de  Tue Sep 18 17:22:36 2012
From: oliver.saggau at sysvision.de (Oliver Saggau | Sysvision GmbH)
Date: Tue, 18 Sep 2012 15:22:36 +0200
Subject: [Freeswitch-users] mod_conference wrong codec for auto_outcall b-leg
Message-ID: <5058759C.9050506@sysvision.de>

Hi everyone,
I got a strange problem with mod_conference and auto_outcall taking the 
wrong codec for b-leg. For what I can see from the log everything seems 
fine. The a-leg calls with the following SDP:

2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP:
v=0
o=- 3556961565 3556961565 IN IP4 10.20.30.125
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 99 101
c=IN IP4 10.20.30.125
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:4001 IN IP4 10.20.30.125

And the SILK/24000 codec is accepted according to the log file:
2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec 
sofia/internal/1000004 at 10.20.30.240 SILK/24000 20 ms 480 samples 40000 bits
2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111 
sofia/internal/1000004 at 10.20.30.240 Original read codec set to SILK:120
2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP 
sofia/internal/1000004 at 10.20.30.240:
v=0
o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240
s=FreeSWITCH
c=IN IP4 10.20.30.240
t=0 0
m=audio 21622 RTP/AVP 99 101
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

The conference get's created and activates the read/write codecs for the 
first member:
2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec 
Activation Success L16 at 24000hz 1 channel 20ms
2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec 
Activation Success L16 at 48000hz 1 channel 20ms

After that he's trying to do the auto_outcall and that's where things 
get weird:
EXECUTE sofia/internal/1000004 at freeswitch.aws.hosts.corpex.de 
conference_set_auto_outcall(sofia/internal/sip:1000003 at 10.20.30.108:63741;transport=UDP;ob)
2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel 
sofia/internal/sip:1000003 at 10.20.30.108:63741 
[0236d005-1ff0-4f13-9370-bf7736087dd0]
2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP:
v=0
o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240
s=FreeSWITCH
c=IN IP4 10.20.30.240
t=0 0
m=audio 20318 RTP/AVP 98 0 8 101 13
a=rtpmap:98 SILK/8000
a=fmtp:98 useinbandfec=1; usedtx=0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec 
Activation Success L16 at 8000hz 1 channel 20ms
2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec 
Activation Success L16 at 48000hz 1 channel 20ms

Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is 
configured like this:
outbound_codec_prefs=silk at 24000,silk at 16000,silk at 8000,PCMU,PCMA,GSM

Thanks for your help,
Oliver


From nasida at live.ru  Tue Sep 18 18:11:18 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Tue, 18 Sep 2012 18:11:18 +0400
Subject: [Freeswitch-users] How to disable adding of rows with LOSE_RACE
	case ?
Message-ID: <SNT127-W56C7748D1F39A7ADF71646D5940@phx.gbl>


Hello guys!
I use many endpoints in my bridge appl. So in cdr I see one good row (for ext which picks up call) and  many rows with "LOSE_RACE" for others ext from bridge string. How can I disable adding of rows with LOSE_RACE ? 
Please advice.Thanks. 		 	   		  
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From avi at avimarcus.net  Tue Sep 18 21:14:01 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 18 Sep 2012 20:14:01 +0300
Subject: [Freeswitch-users] How to disable adding of rows with LOSE_RACE
 case ?
In-Reply-To: <SNT127-W56C7748D1F39A7ADF71646D5940@phx.gbl>
References: <SNT127-W56C7748D1F39A7ADF71646D5940@phx.gbl>
Message-ID: <CALNZuwkaWW9tAz3G8427rfUEcJNFU1_1QUYHG3LvD86gAGDqZQ@mail.gmail.com>

If you use xml_cdr or json_cdr that posts to a database, you can just drop
those CDRs.

I wonder if you can use a hangup hook of a sort (before reporting!) to
set process_cdr to false, or a_only...


-Avi


On Tue, Sep 18, 2012 at 5:11 PM, Yuriy Nasida <nasida at live.ru> wrote:

>  Hello guys!
>
> I use many endpoints in my bridge appl. So in cdr I see one good row (for
> ext which picks up call) and  many rows with "LOSE_RACE" for others ext
> from bridge string. How can I disable adding of rows with LOSE_RACE ?
>
> Please advice.
> 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 msc at freeswitch.org  Tue Sep 18 21:37:53 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 10:37:53 -0700
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <20E48A825239435BB030CC5C30565542@gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
	<CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>
	<CAKzWOxWFgXz5fnDUStz4q53x0C3vRdrR4LFPsY2c9u6f69a2AA@mail.gmail.com>
	<17EE12D5-CC1D-49B4-829B-69EC21111547@freeswitch.org>
	<CAKzWOxUEn9-46qQk2F3-=GonkUQrxfZThZ1-JDrSu-DiZjsicw@mail.gmail.com>
	<20E48A825239435BB030CC5C30565542@gmail.com>
Message-ID: <CAKzWOxVHmM5vSZvYBvc21qgscMJJ5eWERTh1v+JV0cyuTrK93Q@mail.gmail.com>

Seven,

I think you're right about att_xfer being a tool for a different job. I'm
also interested in knowing if using a conference with the mute/unmute and
relations is a feasible approach. However, I think the OP may be able to do
this all with a little bit of scripting and maybe uuid_dual_transfer +
uuid_bridge.

I'll ruminate on this one a bit and maybe we can kick it around after the
featured presentation on tomorrow's conference call.

-MC

On Mon, Sep 17, 2012 at 4:49 PM, Seven Du <dujinfang at gmail.com> wrote:

> I had the similar issue, when using mod_fifo with mixed inbound/outbound
> agents, I have to disable the multiline feature on hard phones (any one
> knows how to disable on eye beam or xlite?)
>
> att_xfer could do this job except it has problems when use with
> event_socket http://jira.freeswitch.org/browse/FS-4419
>  , maybe it's not designed to this kind of usage.
>
> Yes, I would use the similar approach with Alexander if I'm doing this.
> Also I wonder if it's possible to just use a conference with mute/unmute
> hear/nohear and relate/unrelate combinations.
>
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Tuesday, September 18, 2012 at 7:19 AM, Michael Collins wrote:
>
> JM,
>
> That actually explains a lot. I appreciate the information. I can
> definitely see where the relative expense of a hard phone would make this
> kind of application quite valuable. I hope you guys are able to iron out
> the details. Also, maybe you could see if Mitch Capper could add line
> appearances to FSClient. :)
>
> -MC
>
> On Mon, Sep 17, 2012 at 3:42 PM, Jmesquita at freeswitch.org <
> jmesquita at freeswitch.org> wrote:
>
> MC, I believe I have to make a statement here in favor of some realities
> that are different from the ones most people on this list live in. I live
> in South America and an IP Phone here does not get lower than 120usd per
> unit. And I am talking about the cheapest yea link phone model. In brazil,
> where I come from, this is even higher.
>
> Taking that into consideration, several lower end ip phones started to
> appear as well as hybrid systems where the majority of the extensions are
> still analog. In a system like that, CTI applications like the one our
> friend is describing is really the way out to really add value to a
> solution.
>
> Even on the asterisk world we see applications like these. For example the
> flash operator panel and the HUD for asterisk. I believe most of us have
> heard of it before.
>
> Anyhow, I didn't mean to write an essay on the subject but I see this kind
> of feature being constantly rejected by the community and I really can't
> understand why that is.
>
> Jo?o Mesquita
>
> On 17/09/2012, at 01:39 p.m., Michael Collins <msc at freeswitch.org> wrote:
>
> Or get a hard phone that has a hold button and at least two line keys.
> -MC
>
> On Mon, Sep 17, 2012 at 5:19 AM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:
>
> From personal experience, I believe that how you described is the right
> way to do it. The only caveat is that you will have to add variables do the
> channels so you can properly track what is going on in the cdrs . If you
> don't process cdrs, then it is all good. Careful with pickup and such too...
>  On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de>
> wrote:
>
>  Hi MC,****
>
> ** **
>
> thank you for the answer.****
>
> To your question, all call legs in this scenario (outgoing or incoming
> direction) are connected over a sip trunk of a pbx.****
>
> ** **
>
> Scenario:****
>
> My CTI****
>
>   |  A****
>
>   |  |       event socket****
>
>   V  |****
>
> Freeswitch         Sip Trunk****
>
> <-                           PBX****
>
> ->                           ****
>
> ** **
>
> My Client control (CTI) the call legs over the event socket interface and
> the call legs are only legs over the sip trunk to or from the PBX.****
>
> ** **
>
> After some tests i have found a possible solution:****
>
> **-          **Leg A and Leg B are bridged (all legs get the flag
> park_after_bridge = true)****
>
> **-          **For Consultation i park Leg B and transfer this Leg to Moh
> in my default context of my dialplan****
>
> **-          **i originate Leg C and bridge this Leg with Leg A (Leg C
> have the flag park_after_bridge = true too) Consultation is comlete now.**
> **
>
> **-          **with the principle same think i can toggle Leg A <-> Leg B
> and Leg A <-> Leg C.****
>
> ** **
>
> What is your think for this solution? I have tested this on the FS CLI and
> it works.****
>
> ** **
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Michael
> Collins
> *Gesendet:* Freitag, 14. September 2012 19:13
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] Consultation Call via event_socket
> interface****
>
> ** **
>
> Hi Alex,
>
> Welcome to the FreeSWITCH mail list!
>
> First question for you: what kind of telephone are you using? The reason I
> ask is that this kind of function is trivially achieved with a good hard
> phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys.
> If you can use a hard phone w/ multiple line keys then you don't even need
> to mess with the dialplan, uuid_bridge, etc.
>
> -MC****
>
> On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
> wrote:****
>
> Hi All,****
>
>  ****
>
> I?m new on the mailing list.****
>
> I have a problem with a call scenario.****
>
> -          Channel A and channel B are bridged (A is my own channel and B
> is my calling partner)****
>
> -          Now i set channel B on hold with the command ?uuid_hold xxx?
> and create a new channel to C with the command:****
>
> bgapi originate
> {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip
> ',origination_caller_id_number=num at ip}sofia/external/num at ip  &park****
>
> this works correctly, the partner C answer and the channel is established.
> ****
>
> -          Now the Problem:****
>
> I try to bridge the channel a to channel c with the uuid_bridge command,
> now the channel b will hangup, why? The variable hangup_after_bridge is by
> default false.****
>
> Other problem: channel A can hear the voice of channel C but not speak
> with him, channel C can hear and speak. But this problem is not the
> important think at the moment.****
>
> Is there a general problem in my plan to do that?****
>
> Is it a better plan to do this over the dialplan?****
>
> The next step in this scenario is to toggle the connection A -> B and A ->
> C.****
>
>  ****
>
> Thanks for your help!****
>
> Nice regards,****
>
> 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****
>
>
>
>
> --
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Tue Sep 18 21:38:42 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 10:38:42 -0700
Subject: [Freeswitch-users] issue with web server to handle XML CDRs
In-Reply-To: <CAHqn9HF7iZtgdBkPDxdP3QSTAeEEMmqvxUAiocpP_4q1-8=CMA@mail.gmail.com>
References: <CAHqn9HFfo9JiB5mz747Pg18q5hPZXgfV6cnroyQF_SqBkcepzA@mail.gmail.com>
	<CAKzWOxXz5aCKXBFt0a1wFysMUC6_wCzJCC=M9usJqKHPKfgXRw@mail.gmail.com>
	<CAHqn9HF7iZtgdBkPDxdP3QSTAeEEMmqvxUAiocpP_4q1-8=CMA@mail.gmail.com>
Message-ID: <CAKzWOxW3Uj_2MCqHHHx60w_4JKW9TCpjuqDQBDVm9KgYGoRzoQ@mail.gmail.com>

What's in the web server log?

On Mon, Sep 17, 2012 at 7:27 PM, Jagadish Thoutam <jaganthoutam at gmail.com>wrote:

> Hi Michael,
>
>    There are no errors in fs_cli i thing  the request is not reaching to
> web server.
>
> On 17 September 2012 09:37, Michael Collins <msc at freeswitch.org> wrote:
>
>> Next step  is to watch the fs_cli to see if there are any errors. Also,
>> look at your web server access and error logs to see if the request is
>> making it there.
>> -MC
>>
>>  On Mon, Sep 17, 2012 at 2:36 AM, Jagadish Thoutam <
>> jaganthoutam at gmail.com> wrote:
>>
>>>  Hi all,
>>>
>>>    There is the issue with xml_cdr for me when i try to call <param
>>> name="url" value="http://localhost/cgi-bin/cdr.pl"/> its not working, i
>>> have load the mod_xml_cdr on cli, and my cdr.pl file is working well
>>> when try it manually and also have executable permission for cdr.plfile, please some one tell me what i am missing here.
>>>
>>>
>>>
>>> Thanks
>>>
>>> Jagadish
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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.freeswitch.org/>
>> http://www.ClueCon.com <http://www.cluecon.com/>
>> http://www.OSTAG.org <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 msc at freeswitch.org  Tue Sep 18 21:40:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 10:40:14 -0700
Subject: [Freeswitch-users] xml curl
In-Reply-To: <5AE42E9E-D786-442D-A41F-3414F194025C@gmail.com>
References: <50583A50.8020405@softnet.si>
	<5AE42E9E-D786-442D-A41F-3414F194025C@gmail.com>
Message-ID: <CAKzWOxV4O0Nitt5khLA7r8euWKvS-9CH4-8DELUu0ZWoxSPSTA@mail.gmail.com>

This page has good info to start with:
http://wiki.freeswitch.org/wiki/Xml_curl#How_XML_CURL_co-exists_with_static_files

-MC

On Tue, Sep 18, 2012 at 3:33 AM, Nelson Camargo <bigx333 at gmail.com> wrote:

> Yes, if the freeswitch can't find a route in the xml response it will fall
> back to the .xml file.
>
> On 18 Sep 2012, at 11:09 AM, Miha wrote:
>
> > hi,
> >
> > is it possible to use xml_curl just for a one part of my dialplan?
> >
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 klubvps at gmail.com  Tue Sep 18 21:45:07 2012
From: klubvps at gmail.com (siklub)
Date: Wed, 19 Sep 2012 00:45:07 +0700
Subject: [Freeswitch-users] is the example flex client can't receive a call?
Message-ID: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>

Hi All,
I just try the example flex client.
It can make a call to 888 at conference.freeswitch.org
But it can't receive a call.
Some error message, maybe:
switch_ivr_originate.c:2591 Cannot create outgoing channel of type
[error] cause: [USER_NOT_REGISTERED]
mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
....
rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)

Is the example can receive a call? (it is only example, maybe it is incomplete).


From klubvps at gmail.com  Tue Sep 18 21:57:09 2012
From: klubvps at gmail.com (siklub)
Date: Wed, 19 Sep 2012 00:57:09 +0700
Subject: [Freeswitch-users] is the example flex client can't receive a
	call?
In-Reply-To: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
Message-ID: <CAH3d9A=nyvqeeb-6uyryXhKSGCxdc_Wy+LEmmOyf75jfphOVOw@mail.gmail.com>

Complete log message

2012-09-19 00:50:37.328625 [INFO] rtmp_sig.c:136 Replied to createStream (1)
2012-09-19 00:50:37.348525 [WARNING] rtmp.c:100 [amfnumber=2]
Unhandled control packet (type=0x3)
2012-09-19 00:50:37.348525 [INFO] rtmp_sig.c:136 Replied to createStream (2)
2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
rtmp/default/sip:1008 at my.ip.he.re
[5a80e032-01b9-11e2-85d9-ddda9a7c08b2]
2012-09-19 00:50:37.348525 [INFO] mod_dialplan_xml.c:485 Processing
<1007>->sip:1008 at my.ip.he.re in context default
2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
sofia/internal/1008 at my.ip.he.re [5a817632-01b9-11e2-85e3-ddda9a7c08b2]
2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
sofia/internal/1007 at my.ip.he.re [5a81e2c0-01b9-11e2-85e7-ddda9a7c08b2]
2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
Extension 1007 <1007>->1008 in context public
2012-09-19 00:50:37.368624 [NOTICE] switch_ivr.c:1748 Transfer
rtmp/default/sip:1008 at my.ip.he.re to XML[1008 at default]
2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:3326 Hangup
sofia/internal/1008 at my.ip.he.re [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
 Cause: ORIGINATOR_CANCEL
2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
<1007>->1008 in context default
2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
*1 execute_extension::dx XML features
2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
*2 record_session::/usr/local/freeswitch/recordings/1007.2012-09-19-00-50-37.wav
2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
*3 execute_extension::cf XML features
2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
*4 execute_extension::att_xfer XML features
2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
 Cause: USER_NOT_REGISTERED
2012-09-19 00:50:37.368624 [NOTICE] mod_rtmp.c:498 Channel
[rtmp/default/sip:1008 at my.ip.he.re] has been answered
2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1433 Session
2 (sofia/internal/1008 at my.ip.he.re) Ended
2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1437 Close
Channel sofia/internal/1008 at my.ip.he.re [CS_DESTROY]
2012-09-19 00:50:37.468524 [NOTICE] sofia.c:420 Hangup
sofia/internal/1007 at my.ip.he.re [CS_EXECUTE] [BLIND_TRANSFER]
2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1433 Session
3 (sofia/internal/1007 at my.ip.he.re) Ended
2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1437 Close
Channel sofia/internal/1007 at my.ip.he.re [CS_DESTROY]
2012-09-19 00:50:37.948524 [WARNING] rtmp.c:100 [amfnumber=2]
Unhandled control packet (type=0x3)
2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:159 Sending audio
2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:274 Got publish on stream 2.
2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
loopback/app=voicemail:default my.ip.he.re 1008-a
[5b22636c-01b9-11e2-860c-ddda9a7c08b2]
2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:949 Rename
Channel loopback/app=voicemail:default my.ip.he.re
1008-a->loopback/voicemail-a [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
loopback/voicemail-b [5b228cac-01b9-11e2-8610-ddda9a7c08b2]
2012-09-19 00:50:38.408530 [NOTICE] mod_loopback.c:824 Pre-Answer
loopback/voicemail-a!
2012-09-19 00:50:38.408530 [NOTICE] mod_dptools.c:1186 Pre-Answer
loopback/voicemail-b!
2012-09-19 00:50:49.268537 [NOTICE] switch_ivr_play_say.c:399 Channel
[loopback/voicemail-b] has been answered
2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:262
loopback/voicemail-b has executed the last dialplan instruction,
hanging up.
2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:264
Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING]
2012-09-19 00:51:30.468527 [NOTICE] mod_loopback.c:464 Hangup
loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2012-09-19 00:51:30.488529 [NOTICE] switch_ivr_bridge.c:1485 Hangup
rtmp/default/sip:1008 at my.ip.he.re [CS_EXECUTE] [NORMAL_CLEARING]
2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
4 (loopback/voicemail-a) Ended
2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
Channel loopback/voicemail-a [CS_DESTROY]
2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
1 (rtmp/default/sip:1008 at my.ip.he.re) Ended
2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
Channel rtmp/default/sip:1008 at my.ip.he.re [CS_DESTROY]
2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
5 (loopback/voicemail-b) Ended
2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
Channel loopback/voicemail-b [CS_DESTROY]

On Wed, Sep 19, 2012 at 12:45 AM, siklub <klubvps at gmail.com> wrote:
> Hi All,
> I just try the example flex client.
> It can make a call to 888 at conference.freeswitch.org
> But it can't receive a call.
> Some error message, maybe:
> switch_ivr_originate.c:2591 Cannot create outgoing channel of type
> [error] cause: [USER_NOT_REGISTERED]
> mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
> ....
> rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
>
> Is the example can receive a call? (it is only example, maybe it is incomplete).


From msc at freeswitch.org  Tue Sep 18 22:10:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 11:10:27 -0700
Subject: [Freeswitch-users] is the example flex client can't receive a
	call?
In-Reply-To: <CAH3d9A=nyvqeeb-6uyryXhKSGCxdc_Wy+LEmmOyf75jfphOVOw@mail.gmail.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
	<CAH3d9A=nyvqeeb-6uyryXhKSGCxdc_Wy+LEmmOyf75jfphOVOw@mail.gmail.com>
Message-ID: <CAKzWOxXs=pcP8Gewb-Ec1WVCawWmDGZmiLcMksxSw4NOgD+SNA@mail.gmail.com>

I have to ask... is there really a phone registered to 1008?
-MC

On Tue, Sep 18, 2012 at 10:57 AM, siklub <klubvps at gmail.com> wrote:

> Complete log message
>
> 2012-09-19 00:50:37.328625 [INFO] rtmp_sig.c:136 Replied to createStream
> (1)
> 2012-09-19 00:50:37.348525 [WARNING] rtmp.c:100 [amfnumber=2]
> Unhandled control packet (type=0x3)
> 2012-09-19 00:50:37.348525 [INFO] rtmp_sig.c:136 Replied to createStream
> (2)
> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
> rtmp/default/sip:1008 at my.ip.he.re
> [5a80e032-01b9-11e2-85d9-ddda9a7c08b2]
> 2012-09-19 00:50:37.348525 [INFO] mod_dialplan_xml.c:485 Processing
> <1007>->sip:1008 at my.ip.he.re in context default
> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
> sofia/internal/1008 at my.ip.he.re [5a817632-01b9-11e2-85e3-ddda9a7c08b2]
> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
> sofia/internal/1007 at my.ip.he.re [5a81e2c0-01b9-11e2-85e7-ddda9a7c08b2]
> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
> Extension 1007 <1007>->1008 in context public
> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr.c:1748 Transfer
> rtmp/default/sip:1008 at my.ip.he.re to XML[1008 at default]
> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:3326 Hangup
> sofia/internal/1008 at my.ip.he.re [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
>  Cause: ORIGINATOR_CANCEL
> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
> <1007>->1008 in context default
> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> *1 execute_extension::dx XML features
> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> *2
> record_session::/usr/local/freeswitch/recordings/1007.2012-09-19-00-50-37.wav
> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> *3 execute_extension::cf XML features
> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> *4 execute_extension::att_xfer XML features
> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
>  Cause: USER_NOT_REGISTERED
> 2012-09-19 00:50:37.368624 [NOTICE] mod_rtmp.c:498 Channel
> [rtmp/default/sip:1008 at my.ip.he.re] has been answered
> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1433 Session
> 2 (sofia/internal/1008 at my.ip.he.re) Ended
> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1437 Close
> Channel sofia/internal/1008 at my.ip.he.re [CS_DESTROY]
> 2012-09-19 00:50:37.468524 [NOTICE] sofia.c:420 Hangup
> sofia/internal/1007 at my.ip.he.re [CS_EXECUTE] [BLIND_TRANSFER]
> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1433 Session
> 3 (sofia/internal/1007 at my.ip.he.re) Ended
> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1437 Close
> Channel sofia/internal/1007 at my.ip.he.re [CS_DESTROY]
> 2012-09-19 00:50:37.948524 [WARNING] rtmp.c:100 [amfnumber=2]
> Unhandled control packet (type=0x3)
> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:159 Sending audio
> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:274 Got publish on stream 2.
> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
> loopback/app=voicemail:default my.ip.he.re 1008-a
> [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:949 Rename
> Channel loopback/app=voicemail:default my.ip.he.re
> 1008-a->loopback/voicemail-a [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
> loopback/voicemail-b [5b228cac-01b9-11e2-8610-ddda9a7c08b2]
> 2012-09-19 00:50:38.408530 [NOTICE] mod_loopback.c:824 Pre-Answer
> loopback/voicemail-a!
> 2012-09-19 00:50:38.408530 [NOTICE] mod_dptools.c:1186 Pre-Answer
> loopback/voicemail-b!
> 2012-09-19 00:50:49.268537 [NOTICE] switch_ivr_play_say.c:399 Channel
> [loopback/voicemail-b] has been answered
> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:262
> loopback/voicemail-b has executed the last dialplan instruction,
> hanging up.
> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:264
> Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING]
> 2012-09-19 00:51:30.468527 [NOTICE] mod_loopback.c:464 Hangup
> loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2012-09-19 00:51:30.488529 [NOTICE] switch_ivr_bridge.c:1485 Hangup
> rtmp/default/sip:1008 at my.ip.he.re [CS_EXECUTE] [NORMAL_CLEARING]
> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
> 4 (loopback/voicemail-a) Ended
> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
> Channel loopback/voicemail-a [CS_DESTROY]
> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
> 1 (rtmp/default/sip:1008 at my.ip.he.re) Ended
> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
> Channel rtmp/default/sip:1008 at my.ip.he.re [CS_DESTROY]
> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
> 5 (loopback/voicemail-b) Ended
> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
> Channel loopback/voicemail-b [CS_DESTROY]
>
> On Wed, Sep 19, 2012 at 12:45 AM, siklub <klubvps at gmail.com> wrote:
> > Hi All,
> > I just try the example flex client.
> > It can make a call to 888 at conference.freeswitch.org
> > But it can't receive a call.
> > Some error message, maybe:
> > switch_ivr_originate.c:2591 Cannot create outgoing channel of type
> > [error] cause: [USER_NOT_REGISTERED]
> > mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
> > ....
> > rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
> >
> > Is the example can receive a call? (it is only example, maybe it is
> incomplete).
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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> 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 klubvps at gmail.com  Tue Sep 18 22:34:14 2012
From: klubvps at gmail.com (siklub)
Date: Wed, 19 Sep 2012 01:34:14 +0700
Subject: [Freeswitch-users] is the example flex client can't receive a
	call?
In-Reply-To: <CAKzWOxXs=pcP8Gewb-Ec1WVCawWmDGZmiLcMksxSw4NOgD+SNA@mail.gmail.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
	<CAH3d9A=nyvqeeb-6uyryXhKSGCxdc_Wy+LEmmOyf75jfphOVOw@mail.gmail.com>
	<CAKzWOxXs=pcP8Gewb-Ec1WVCawWmDGZmiLcMksxSw4NOgD+SNA@mail.gmail.com>
Message-ID: <CAH3d9AkMv3D+fHt1OER6tzXi0Fhar4okhSv7iVydK6QckG6-Mg@mail.gmail.com>

HI michael, thanks for quick response.
I login as 1007 using the example flex client in a computer.
And login as 1008 using the example flex client from another computer.

On Wed, Sep 19, 2012 at 1:10 AM, Michael Collins <msc at freeswitch.org> wrote:
> I have to ask... is there really a phone registered to 1008?
> -MC
>
> On Tue, Sep 18, 2012 at 10:57 AM, siklub <klubvps at gmail.com> wrote:
>>
>> Complete log message
>>
>> 2012-09-19 00:50:37.328625 [INFO] rtmp_sig.c:136 Replied to createStream
>> (1)
>> 2012-09-19 00:50:37.348525 [WARNING] rtmp.c:100 [amfnumber=2]
>> Unhandled control packet (type=0x3)
>> 2012-09-19 00:50:37.348525 [INFO] rtmp_sig.c:136 Replied to createStream
>> (2)
>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
>> rtmp/default/sip:1008 at my.ip.he.re
>> [5a80e032-01b9-11e2-85d9-ddda9a7c08b2]
>> 2012-09-19 00:50:37.348525 [INFO] mod_dialplan_xml.c:485 Processing
>> <1007>->sip:1008 at my.ip.he.re in context default
>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
>> sofia/internal/1008 at my.ip.he.re [5a817632-01b9-11e2-85e3-ddda9a7c08b2]
>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
>> sofia/internal/1007 at my.ip.he.re [5a81e2c0-01b9-11e2-85e7-ddda9a7c08b2]
>> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
>> Extension 1007 <1007>->1008 in context public
>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr.c:1748 Transfer
>> rtmp/default/sip:1008 at my.ip.he.re to XML[1008 at default]
>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:3326 Hangup
>> sofia/internal/1008 at my.ip.he.re [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
>>  Cause: ORIGINATOR_CANCEL
>> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
>> <1007>->1008 in context default
>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>> *1 execute_extension::dx XML features
>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>> *2
>> record_session::/usr/local/freeswitch/recordings/1007.2012-09-19-00-50-37.wav
>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>> *3 execute_extension::cf XML features
>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>> *4 execute_extension::att_xfer XML features
>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
>>  Cause: USER_NOT_REGISTERED
>> 2012-09-19 00:50:37.368624 [NOTICE] mod_rtmp.c:498 Channel
>> [rtmp/default/sip:1008 at my.ip.he.re] has been answered
>> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1433 Session
>> 2 (sofia/internal/1008 at my.ip.he.re) Ended
>> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1437 Close
>> Channel sofia/internal/1008 at my.ip.he.re [CS_DESTROY]
>> 2012-09-19 00:50:37.468524 [NOTICE] sofia.c:420 Hangup
>> sofia/internal/1007 at my.ip.he.re [CS_EXECUTE] [BLIND_TRANSFER]
>> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1433 Session
>> 3 (sofia/internal/1007 at my.ip.he.re) Ended
>> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1437 Close
>> Channel sofia/internal/1007 at my.ip.he.re [CS_DESTROY]
>> 2012-09-19 00:50:37.948524 [WARNING] rtmp.c:100 [amfnumber=2]
>> Unhandled control packet (type=0x3)
>> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:159 Sending audio
>> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:274 Got publish on stream 2.
>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
>> loopback/app=voicemail:default my.ip.he.re 1008-a
>> [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:949 Rename
>> Channel loopback/app=voicemail:default my.ip.he.re
>> 1008-a->loopback/voicemail-a [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
>> loopback/voicemail-b [5b228cac-01b9-11e2-8610-ddda9a7c08b2]
>> 2012-09-19 00:50:38.408530 [NOTICE] mod_loopback.c:824 Pre-Answer
>> loopback/voicemail-a!
>> 2012-09-19 00:50:38.408530 [NOTICE] mod_dptools.c:1186 Pre-Answer
>> loopback/voicemail-b!
>> 2012-09-19 00:50:49.268537 [NOTICE] switch_ivr_play_say.c:399 Channel
>> [loopback/voicemail-b] has been answered
>> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:262
>> loopback/voicemail-b has executed the last dialplan instruction,
>> hanging up.
>> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:264
>> Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING]
>> 2012-09-19 00:51:30.468527 [NOTICE] mod_loopback.c:464 Hangup
>> loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> 2012-09-19 00:51:30.488529 [NOTICE] switch_ivr_bridge.c:1485 Hangup
>> rtmp/default/sip:1008 at my.ip.he.re [CS_EXECUTE] [NORMAL_CLEARING]
>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
>> 4 (loopback/voicemail-a) Ended
>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
>> Channel loopback/voicemail-a [CS_DESTROY]
>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
>> 1 (rtmp/default/sip:1008 at my.ip.he.re) Ended
>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
>> Channel rtmp/default/sip:1008 at my.ip.he.re [CS_DESTROY]
>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
>> 5 (loopback/voicemail-b) Ended
>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
>> Channel loopback/voicemail-b [CS_DESTROY]
>>
>> On Wed, Sep 19, 2012 at 12:45 AM, siklub <klubvps at gmail.com> wrote:
>> > Hi All,
>> > I just try the example flex client.
>> > It can make a call to 888 at conference.freeswitch.org
>> > But it can't receive a call.
>> > Some error message, maybe:
>> > switch_ivr_originate.c:2591 Cannot create outgoing channel of type
>> > [error] cause: [USER_NOT_REGISTERED]
>> > mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
>> > ....
>> > rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
>> >
>> > Is the example can receive a call? (it is only example, maybe it is
>> > incomplete).
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 klubvps at gmail.com  Tue Sep 18 23:58:26 2012
From: klubvps at gmail.com (siklub)
Date: Wed, 19 Sep 2012 02:58:26 +0700
Subject: [Freeswitch-users] is the example flex client can't receive a
	call?
In-Reply-To: <CAH3d9AkMv3D+fHt1OER6tzXi0Fhar4okhSv7iVydK6QckG6-Mg@mail.gmail.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
	<CAH3d9A=nyvqeeb-6uyryXhKSGCxdc_Wy+LEmmOyf75jfphOVOw@mail.gmail.com>
	<CAKzWOxXs=pcP8Gewb-Ec1WVCawWmDGZmiLcMksxSw4NOgD+SNA@mail.gmail.com>
	<CAH3d9AkMv3D+fHt1OER6tzXi0Fhar4okhSv7iVydK6QckG6-Mg@mail.gmail.com>
Message-ID: <CAH3d9Ak0sR7KXebWrdJkwhbxb6SKS7YfSOW4PxBMihTDduT6tA@mail.gmail.com>

additional info.
i can make a call from flex client to x-lite.

On Wed, Sep 19, 2012 at 1:34 AM, siklub <klubvps at gmail.com> wrote:
> HI michael, thanks for quick response.
> I login as 1007 using the example flex client in a computer.
> And login as 1008 using the example flex client from another computer.
>
> On Wed, Sep 19, 2012 at 1:10 AM, Michael Collins <msc at freeswitch.org> wrote:
>> I have to ask... is there really a phone registered to 1008?
>> -MC
>>
>> On Tue, Sep 18, 2012 at 10:57 AM, siklub <klubvps at gmail.com> wrote:
>>>
>>> Complete log message
>>>
>>> 2012-09-19 00:50:37.328625 [INFO] rtmp_sig.c:136 Replied to createStream
>>> (1)
>>> 2012-09-19 00:50:37.348525 [WARNING] rtmp.c:100 [amfnumber=2]
>>> Unhandled control packet (type=0x3)
>>> 2012-09-19 00:50:37.348525 [INFO] rtmp_sig.c:136 Replied to createStream
>>> (2)
>>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
>>> rtmp/default/sip:1008 at my.ip.he.re
>>> [5a80e032-01b9-11e2-85d9-ddda9a7c08b2]
>>> 2012-09-19 00:50:37.348525 [INFO] mod_dialplan_xml.c:485 Processing
>>> <1007>->sip:1008 at my.ip.he.re in context default
>>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
>>> sofia/internal/1008 at my.ip.he.re [5a817632-01b9-11e2-85e3-ddda9a7c08b2]
>>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
>>> sofia/internal/1007 at my.ip.he.re [5a81e2c0-01b9-11e2-85e7-ddda9a7c08b2]
>>> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
>>> Extension 1007 <1007>->1008 in context public
>>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr.c:1748 Transfer
>>> rtmp/default/sip:1008 at my.ip.he.re to XML[1008 at default]
>>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:3326 Hangup
>>> sofia/internal/1008 at my.ip.he.re [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
>>>  Cause: ORIGINATOR_CANCEL
>>> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
>>> <1007>->1008 in context default
>>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>>> *1 execute_extension::dx XML features
>>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>>> *2
>>> record_session::/usr/local/freeswitch/recordings/1007.2012-09-19-00-50-37.wav
>>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>>> *3 execute_extension::cf XML features
>>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
>>> *4 execute_extension::att_xfer XML features
>>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>>> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
>>>  Cause: USER_NOT_REGISTERED
>>> 2012-09-19 00:50:37.368624 [NOTICE] mod_rtmp.c:498 Channel
>>> [rtmp/default/sip:1008 at my.ip.he.re] has been answered
>>> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1433 Session
>>> 2 (sofia/internal/1008 at my.ip.he.re) Ended
>>> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1437 Close
>>> Channel sofia/internal/1008 at my.ip.he.re [CS_DESTROY]
>>> 2012-09-19 00:50:37.468524 [NOTICE] sofia.c:420 Hangup
>>> sofia/internal/1007 at my.ip.he.re [CS_EXECUTE] [BLIND_TRANSFER]
>>> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1433 Session
>>> 3 (sofia/internal/1007 at my.ip.he.re) Ended
>>> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1437 Close
>>> Channel sofia/internal/1007 at my.ip.he.re [CS_DESTROY]
>>> 2012-09-19 00:50:37.948524 [WARNING] rtmp.c:100 [amfnumber=2]
>>> Unhandled control packet (type=0x3)
>>> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:159 Sending audio
>>> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:274 Got publish on stream 2.
>>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
>>> loopback/app=voicemail:default my.ip.he.re 1008-a
>>> [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
>>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:949 Rename
>>> Channel loopback/app=voicemail:default my.ip.he.re
>>> 1008-a->loopback/voicemail-a [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
>>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
>>> loopback/voicemail-b [5b228cac-01b9-11e2-8610-ddda9a7c08b2]
>>> 2012-09-19 00:50:38.408530 [NOTICE] mod_loopback.c:824 Pre-Answer
>>> loopback/voicemail-a!
>>> 2012-09-19 00:50:38.408530 [NOTICE] mod_dptools.c:1186 Pre-Answer
>>> loopback/voicemail-b!
>>> 2012-09-19 00:50:49.268537 [NOTICE] switch_ivr_play_say.c:399 Channel
>>> [loopback/voicemail-b] has been answered
>>> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:262
>>> loopback/voicemail-b has executed the last dialplan instruction,
>>> hanging up.
>>> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:264
>>> Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2012-09-19 00:51:30.468527 [NOTICE] mod_loopback.c:464 Hangup
>>> loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>> 2012-09-19 00:51:30.488529 [NOTICE] switch_ivr_bridge.c:1485 Hangup
>>> rtmp/default/sip:1008 at my.ip.he.re [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
>>> 4 (loopback/voicemail-a) Ended
>>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
>>> Channel loopback/voicemail-a [CS_DESTROY]
>>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
>>> 1 (rtmp/default/sip:1008 at my.ip.he.re) Ended
>>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
>>> Channel rtmp/default/sip:1008 at my.ip.he.re [CS_DESTROY]
>>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
>>> 5 (loopback/voicemail-b) Ended
>>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
>>> Channel loopback/voicemail-b [CS_DESTROY]
>>>
>>> On Wed, Sep 19, 2012 at 12:45 AM, siklub <klubvps at gmail.com> wrote:
>>> > Hi All,
>>> > I just try the example flex client.
>>> > It can make a call to 888 at conference.freeswitch.org
>>> > But it can't receive a call.
>>> > Some error message, maybe:
>>> > switch_ivr_originate.c:2591 Cannot create outgoing channel of type
>>> > [error] cause: [USER_NOT_REGISTERED]
>>> > mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
>>> > ....
>>> > rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
>>> >
>>> > Is the example can receive a call? (it is only example, maybe it is
>>> > incomplete).
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 msc at freeswitch.org  Wed Sep 19 00:52:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 13:52:41 -0700
Subject: [Freeswitch-users] is the example flex client can't receive a
	call?
In-Reply-To: <CAH3d9Ak0sR7KXebWrdJkwhbxb6SKS7YfSOW4PxBMihTDduT6tA@mail.gmail.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
	<CAH3d9A=nyvqeeb-6uyryXhKSGCxdc_Wy+LEmmOyf75jfphOVOw@mail.gmail.com>
	<CAKzWOxXs=pcP8Gewb-Ec1WVCawWmDGZmiLcMksxSw4NOgD+SNA@mail.gmail.com>
	<CAH3d9AkMv3D+fHt1OER6tzXi0Fhar4okhSv7iVydK6QckG6-Mg@mail.gmail.com>
	<CAH3d9Ak0sR7KXebWrdJkwhbxb6SKS7YfSOW4PxBMihTDduT6tA@mail.gmail.com>
Message-ID: <CAKzWOxVfpWY8FdNm5hOBwL1c32uy++QUbopy=Q33A9QRiF0v1A@mail.gmail.com>

Okay, it looks like 1008 is your flex client, but if you're using the
example configs then the dialplan for 1008 is going to look for SIP user
1008. Maybe create a different user?
-MC

On Tue, Sep 18, 2012 at 12:58 PM, siklub <klubvps at gmail.com> wrote:

> additional info.
> i can make a call from flex client to x-lite.
>
> On Wed, Sep 19, 2012 at 1:34 AM, siklub <klubvps at gmail.com> wrote:
> > HI michael, thanks for quick response.
> > I login as 1007 using the example flex client in a computer.
> > And login as 1008 using the example flex client from another computer.
> >
> > On Wed, Sep 19, 2012 at 1:10 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> >> I have to ask... is there really a phone registered to 1008?
> >> -MC
> >>
> >> On Tue, Sep 18, 2012 at 10:57 AM, siklub <klubvps at gmail.com> wrote:
> >>>
> >>> Complete log message
> >>>
> >>> 2012-09-19 00:50:37.328625 [INFO] rtmp_sig.c:136 Replied to
> createStream
> >>> (1)
> >>> 2012-09-19 00:50:37.348525 [WARNING] rtmp.c:100 [amfnumber=2]
> >>> Unhandled control packet (type=0x3)
> >>> 2012-09-19 00:50:37.348525 [INFO] rtmp_sig.c:136 Replied to
> createStream
> >>> (2)
> >>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
> >>> rtmp/default/sip:1008 at my.ip.he.re
> >>> [5a80e032-01b9-11e2-85d9-ddda9a7c08b2]
> >>> 2012-09-19 00:50:37.348525 [INFO] mod_dialplan_xml.c:485 Processing
> >>> <1007>->sip:1008 at my.ip.he.re in context default
> >>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
> >>> sofia/internal/1008 at my.ip.he.re [5a817632-01b9-11e2-85e3-ddda9a7c08b2]
> >>> 2012-09-19 00:50:37.348525 [NOTICE] switch_channel.c:951 New Channel
> >>> sofia/internal/1007 at my.ip.he.re [5a81e2c0-01b9-11e2-85e7-ddda9a7c08b2]
> >>> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
> >>> Extension 1007 <1007>->1008 in context public
> >>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr.c:1748 Transfer
> >>> rtmp/default/sip:1008 at my.ip.he.re to XML[1008 at default]
> >>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:3326 Hangup
> >>> sofia/internal/1008 at my.ip.he.re [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> >>> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
> >>>  Cause: ORIGINATOR_CANCEL
> >>> 2012-09-19 00:50:37.368624 [INFO] mod_dialplan_xml.c:485 Processing
> >>> <1007>->1008 in context default
> >>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> >>> *1 execute_extension::dx XML features
> >>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> >>> *2
> >>>
> record_session::/usr/local/freeswitch/recordings/1007.2012-09-19-00-50-37.wav
> >>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> >>> *3 execute_extension::cf XML features
> >>> 2012-09-19 00:50:37.368624 [INFO] switch_ivr_async.c:3357 Bound B-Leg:
> >>> *4 execute_extension::att_xfer XML features
> >>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
> >>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> >>> 2012-09-19 00:50:37.368624 [NOTICE] switch_ivr_originate.c:2591 Cannot
> >>> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> >>> 2012-09-19 00:50:37.368624 [INFO] mod_dptools.c:3027 Originate Failed.
> >>>  Cause: USER_NOT_REGISTERED
> >>> 2012-09-19 00:50:37.368624 [NOTICE] mod_rtmp.c:498 Channel
> >>> [rtmp/default/sip:1008 at my.ip.he.re] has been answered
> >>> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1433 Session
> >>> 2 (sofia/internal/1008 at my.ip.he.re) Ended
> >>> 2012-09-19 00:50:37.388534 [NOTICE] switch_core_session.c:1437 Close
> >>> Channel sofia/internal/1008 at my.ip.he.re [CS_DESTROY]
> >>> 2012-09-19 00:50:37.468524 [NOTICE] sofia.c:420 Hangup
> >>> sofia/internal/1007 at my.ip.he.re [CS_EXECUTE] [BLIND_TRANSFER]
> >>> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1433 Session
> >>> 3 (sofia/internal/1007 at my.ip.he.re) Ended
> >>> 2012-09-19 00:50:37.468524 [NOTICE] switch_core_session.c:1437 Close
> >>> Channel sofia/internal/1007 at my.ip.he.re [CS_DESTROY]
> >>> 2012-09-19 00:50:37.948524 [WARNING] rtmp.c:100 [amfnumber=2]
> >>> Unhandled control packet (type=0x3)
> >>> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:159 Sending audio
> >>> 2012-09-19 00:50:37.948524 [INFO] rtmp_sig.c:274 Got publish on stream
> 2.
> >>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
> >>> loopback/app=voicemail:default my.ip.he.re 1008-a
> >>> [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
> >>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:949 Rename
> >>> Channel loopback/app=voicemail:default my.ip.he.re
> >>> 1008-a->loopback/voicemail-a [5b22636c-01b9-11e2-860c-ddda9a7c08b2]
> >>> 2012-09-19 00:50:38.408530 [NOTICE] switch_channel.c:951 New Channel
> >>> loopback/voicemail-b [5b228cac-01b9-11e2-8610-ddda9a7c08b2]
> >>> 2012-09-19 00:50:38.408530 [NOTICE] mod_loopback.c:824 Pre-Answer
> >>> loopback/voicemail-a!
> >>> 2012-09-19 00:50:38.408530 [NOTICE] mod_dptools.c:1186 Pre-Answer
> >>> loopback/voicemail-b!
> >>> 2012-09-19 00:50:49.268537 [NOTICE] switch_ivr_play_say.c:399 Channel
> >>> [loopback/voicemail-b] has been answered
> >>> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:262
> >>> loopback/voicemail-b has executed the last dialplan instruction,
> >>> hanging up.
> >>> 2012-09-19 00:51:30.468527 [NOTICE] switch_core_state_machine.c:264
> >>> Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING]
> >>> 2012-09-19 00:51:30.468527 [NOTICE] mod_loopback.c:464 Hangup
> >>> loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> >>> 2012-09-19 00:51:30.488529 [NOTICE] switch_ivr_bridge.c:1485 Hangup
> >>> rtmp/default/sip:1008 at my.ip.he.re [CS_EXECUTE] [NORMAL_CLEARING]
> >>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
> >>> 4 (loopback/voicemail-a) Ended
> >>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
> >>> Channel loopback/voicemail-a [CS_DESTROY]
> >>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
> >>> 1 (rtmp/default/sip:1008 at my.ip.he.re) Ended
> >>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
> >>> Channel rtmp/default/sip:1008 at my.ip.he.re [CS_DESTROY]
> >>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1433 Session
> >>> 5 (loopback/voicemail-b) Ended
> >>> 2012-09-19 00:51:30.488529 [NOTICE] switch_core_session.c:1437 Close
> >>> Channel loopback/voicemail-b [CS_DESTROY]
> >>>
> >>> On Wed, Sep 19, 2012 at 12:45 AM, siklub <klubvps at gmail.com> wrote:
> >>> > Hi All,
> >>> > I just try the example flex client.
> >>> > It can make a call to 888 at conference.freeswitch.org
> >>> > But it can't receive a call.
> >>> > Some error message, maybe:
> >>> > switch_ivr_originate.c:2591 Cannot create outgoing channel of type
> >>> > [error] cause: [USER_NOT_REGISTERED]
> >>> > mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
> >>> > ....
> >>> > rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
> >>> >
> >>> > Is the example can receive a call? (it is only example, maybe it is
> >>> > incomplete).
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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 georg at riseup.net  Wed Sep 19 00:53:39 2012
From: georg at riseup.net (georg at riseup.net)
Date: Tue, 18 Sep 2012 22:53:39 +0200
Subject: [Freeswitch-users] Strip digits from incoming caller id
Message-ID: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>

Hi all,

My provider is sending me the phone numbers in the E.164 (I guess) format.
I would like to remove the international prefix (on local fixed line and
local mobile calls at least), and add a leading zero. I know how to add
digits, but I've got problems to remove digits.

My dialplan at the moment looks like this:

<extension name="Dispo Intern" continue="true">
 <condition field="destination_number" expression="^(123456789)$">
  <regex field="caller_id_number" expression="^49(\d+)$2"/>
  <action application="set" data="effective_caller_id_number=0$2"/>
  <action application="set" data="effective_caller_id_name=_undef_"/>
  <anti action application="set"
data="effective_caller_id_number=${original_caller_id_number}"/>
 </condition>
</extension>

What I'm trying to do:
- Check if the call is a "local" one
- If yes, alter the caller id
- I thought, I'm able to save the stripped caller id in $2, and use it
again in the next line; this is not true, or at least not in the way I did
this
- It seems that $2 is "empty", because the altered caller id is just 0

Could someone point me into the right direction?

Thanks in advance,
Georg



From paul at cupis.co.uk  Wed Sep 19 01:08:25 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Tue, 18 Sep 2012 22:08:25 +0100
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
Message-ID: <5058E2C9.2080308@cupis.co.uk>

On 18/09/12 21:53, georg at riseup.net wrote:
> My provider is sending me the phone numbers in the E.164 (I guess) format.
> I would like to remove the international prefix (on local fixed line and
> local mobile calls at least), and add a leading zero. I know how to add
> digits, but I've got problems to remove digits.

>    <regex field="caller_id_number" expression="^49(\d+)$2"/>
>    <action application="set" data="effective_caller_id_number=0$2"/>

Try this instead of the above:

<action application="set" 
data="effective_caller_id_number=${regex(${caller_id_number}|^49([0-9]{9})$|0%1)}"/>

You maybe want to adjust the {9} to reflect the correct number of digits 
in the national part of the caller ID.

The above is derived from:

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

Regards,


From georg at riseup.net  Wed Sep 19 01:20:13 2012
From: georg at riseup.net (georg at riseup.net)
Date: Tue, 18 Sep 2012 23:20:13 +0200
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <5058E2C9.2080308@cupis.co.uk>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
	<5058E2C9.2080308@cupis.co.uk>
Message-ID: <65d848bade245bbb25d895325f570d28.squirrel@fruiteater.riseup.net>

Hi Paul,

> Try this instead of the above:
>
> <action application="set"
> data="effective_caller_id_number=${regex(${caller_id_number}|^49([0-9]{9})$|0%1)}"/>
>
> You maybe want to adjust the {9} to reflect the correct number of digits
> in the national part of the caller ID.

Thank you. I've found this aswell in the wiki 2 minutes after my post, and
wanted to test first.
Everything works as expected! :)

Cheers,
Georg

P.S.: Before my post I googled aroung for sth. like two hours. Should I
wikify this somewhere (if yes, where?) to make it easier to find?



From msc at freeswitch.org  Wed Sep 19 01:20:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 14:20:04 -0700
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
Message-ID: <CAKzWOxX=v2am2QOPwi7VjbxtrWu0XwB91e+f7anusJtBmOX9Pw@mail.gmail.com>

Georg,

If you just want the phone number w/o the international prefix then it's
better to capture that correctly the first time:
<condition field="destination_number" expression="^(49)?(123456789)$">
The value you want is now in $2, and if it's international then you'll have
"49" in $1. However, you cannot use $1 or  $2 in another condition. You'll
need to store them before you do anything else:

<extension name="Dispo Intern" continue="true">
  <condition field="destination_number" expression="^(49)?(123456789)$">
    <action application="set" data="international=$1" inline="true"/>
    <action application="set" data="dialed_number=$2" inline="true"/>
  </condition>

  <condition field="${international}" data="^49$">
    <action application="set"
data="effective_caller_id_number=0${dialed_number}"/>
    <action application="set" data="effective_caller_id_name=_undef_"/>
    <anti-action application="set"
data="effective_caller_id_number=${dialed_number}"/>
  </condition>

</extension>

Try that. Standard disclaimer applies: I did that off the top of my head,
so if it doesn't work the first time please test and tinker a bit before
you report back. :)

-MC

On Tue, Sep 18, 2012 at 1:53 PM, <georg at riseup.net> wrote:

> Hi all,
>
> My provider is sending me the phone numbers in the E.164 (I guess) format.
> I would like to remove the international prefix (on local fixed line and
> local mobile calls at least), and add a leading zero. I know how to add
> digits, but I've got problems to remove digits.
>
> My dialplan at the moment looks like this:
>
> <extension name="Dispo Intern" continue="true">
>  <condition field="destination_number" expression="^(123456789)$">
>   <regex field="caller_id_number" expression="^49(\d+)$2"/>
>   <action application="set" data="effective_caller_id_number=0$2"/>
>   <action application="set" data="effective_caller_id_name=_undef_"/>
>   <anti action application="set"
> data="effective_caller_id_number=${original_caller_id_number}"/>
>  </condition>
> </extension>
>
> What I'm trying to do:
> - Check if the call is a "local" one
> - If yes, alter the caller id
> - I thought, I'm able to save the stripped caller id in $2, and use it
> again in the next line; this is not true, or at least not in the way I did
> this
> - It seems that $2 is "empty", because the altered caller id is just 0
>
> Could someone point me into the right direction?
>
> Thanks in advance,
> Georg
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 georg at riseup.net  Wed Sep 19 01:28:20 2012
From: georg at riseup.net (georg at riseup.net)
Date: Tue, 18 Sep 2012 23:28:20 +0200
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <CAKzWOxX=v2am2QOPwi7VjbxtrWu0XwB91e+f7anusJtBmOX9Pw@mail.gmail.com>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
	<CAKzWOxX=v2am2QOPwi7VjbxtrWu0XwB91e+f7anusJtBmOX9Pw@mail.gmail.com>
Message-ID: <00a24f1b68c668d4b619a7c1b94e4ae6.squirrel@fruiteater.riseup.net>

Hi Michael,

> If you just want the phone number w/o the international prefix then it's
> better to capture that correctly the first time:
> <condition field="destination_number" expression="^(49)?(123456789)$">

Yes. Problem is that the number after expression is not the incoming
caller id (which I would like to alter), but instead a DID. I tried to
circumcent this, but with no luck.

To clarify this: This number is not the not the number I would like have
shown on my phones. It's my own number, not the number the calling party
is calling from.

Anyway, the other way round is working great.

Thank you,
Georg



From msc at freeswitch.org  Wed Sep 19 01:33:02 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 14:33:02 -0700
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <65d848bade245bbb25d895325f570d28.squirrel@fruiteater.riseup.net>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
	<5058E2C9.2080308@cupis.co.uk>
	<65d848bade245bbb25d895325f570d28.squirrel@fruiteater.riseup.net>
Message-ID: <CAKzWOxX2BydfAaTL-D80vtRBgbF-zw+9sEYwRyg5m-+y8nbcLg@mail.gmail.com>

Where on the wiki were you looking for this information? It sounds like it
is related to caller id handling. My guess is that the best place to put it
would be:
http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number

if you put it inside of the <onlyinclude> tags then it will also display on
the main channel variables page.

-MC

On Tue, Sep 18, 2012 at 2:20 PM, <georg at riseup.net> wrote:

> Hi Paul,
>
> > Try this instead of the above:
> >
> > <action application="set"
> >
> data="effective_caller_id_number=${regex(${caller_id_number}|^49([0-9]{9})$|0%1)}"/>
> >
> > You maybe want to adjust the {9} to reflect the correct number of digits
> > in the national part of the caller ID.
>
> Thank you. I've found this aswell in the wiki 2 minutes after my post, and
> wanted to test first.
> Everything works as expected! :)
>
> Cheers,
> Georg
>
> P.S.: Before my post I googled aroung for sth. like two hours. Should I
> wikify this somewhere (if yes, where?) to make it easier to find?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Sep 19 01:45:47 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Sep 2012 00:45:47 +0300
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <CAKzWOxX2BydfAaTL-D80vtRBgbF-zw+9sEYwRyg5m-+y8nbcLg@mail.gmail.com>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
	<5058E2C9.2080308@cupis.co.uk>
	<65d848bade245bbb25d895325f570d28.squirrel@fruiteater.riseup.net>
	<CAKzWOxX2BydfAaTL-D80vtRBgbF-zw+9sEYwRyg5m-+y8nbcLg@mail.gmail.com>
Message-ID: <CALNZuwmptSiyvAnVBnJwOxRWpcOxfLD096wC7YzKtHPsL74-kQ@mail.gmail.com>

Whoa. I did NOT know about this onlyinclude stuff.
It looks like there's quite a few vars on channel vars page that aren't
using an include like this.

-Avi


On Wed, Sep 19, 2012 at 12:33 AM, Michael Collins <msc at freeswitch.org>wrote:

> Where on the wiki were you looking for this information? It sounds like it
> is related to caller id handling. My guess is that the best place to put it
> would be:
> http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
>
> if you put it inside of the <onlyinclude> tags then it will also display
> on the main channel variables page.
>
> -MC
>
>
> On Tue, Sep 18, 2012 at 2:20 PM, <georg at riseup.net> wrote:
>
>> Hi Paul,
>>
>> > Try this instead of the above:
>> >
>> > <action application="set"
>> >
>> data="effective_caller_id_number=${regex(${caller_id_number}|^49([0-9]{9})$|0%1)}"/>
>> >
>> > You maybe want to adjust the {9} to reflect the correct number of digits
>> > in the national part of the caller ID.
>>
>> Thank you. I've found this aswell in the wiki 2 minutes after my post, and
>> wanted to test first.
>> Everything works as expected! :)
>>
>> Cheers,
>> Georg
>>
>> P.S.: Before my post I googled aroung for sth. like two hours. Should I
>> wikify this somewhere (if yes, where?) to make it easier to find?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 georg at riseup.net  Wed Sep 19 01:46:22 2012
From: georg at riseup.net (georg at riseup.net)
Date: Tue, 18 Sep 2012 23:46:22 +0200
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <CAKzWOxX2BydfAaTL-D80vtRBgbF-zw+9sEYwRyg5m-+y8nbcLg@mail.gmail.com>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
	<5058E2C9.2080308@cupis.co.uk>
	<65d848bade245bbb25d895325f570d28.squirrel@fruiteater.riseup.net>
	<CAKzWOxX2BydfAaTL-D80vtRBgbF-zw+9sEYwRyg5m-+y8nbcLg@mail.gmail.com>
Message-ID: <80eb858fe62119a6c53d369db8d61893.squirrel@fruiteater.riseup.net>

Hi,

> Where on the wiki were you looking for this information? It sounds like it
> is related to caller id handling. My guess is that the best place to put
> it
> would be:
> http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number

Allright, will put it there.
Is it undesired to store the same information twice? Because if not, I
would also add it to the dialplan examples, maybe next to [1].

Cheers,
Georg

[1]
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_3:_Stripping_leading_digits

> if you put it inside of the <onlyinclude> tags then it will also display
> on
> the main channel variables page.
>
> -MC
>
> On Tue, Sep 18, 2012 at 2:20 PM, <georg at riseup.net> wrote:
>
>> Hi Paul,
>>
>> > Try this instead of the above:
>> >
>> > <action application="set"
>> >
>> data="effective_caller_id_number=${regex(${caller_id_number}|^49([0-9]{9})$|0%1)}"/>
>> >
>> > You maybe want to adjust the {9} to reflect the correct number of
>> digits
>> > in the national part of the caller ID.
>>
>> Thank you. I've found this aswell in the wiki 2 minutes after my post,
>> and
>> wanted to test first.
>> Everything works as expected! :)
>>
>> Cheers,
>> Georg
>>
>> P.S.: Before my post I googled aroung for sth. like two hours. Should I
>> wikify this somewhere (if yes, where?) to make it easier to find?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 nasida at live.ru  Wed Sep 19 01:51:05 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 19 Sep 2012 01:51:05 +0400
Subject: [Freeswitch-users] How to disable adding of rows with LOSE_RACE
 case ?
In-Reply-To: <CALNZuwkaWW9tAz3G8427rfUEcJNFU1_1QUYHG3LvD86gAGDqZQ@mail.gmail.com>
References: <SNT127-W56C7748D1F39A7ADF71646D5940@phx.gbl>,
	<CALNZuwkaWW9tAz3G8427rfUEcJNFU1_1QUYHG3LvD86gAGDqZQ@mail.gmail.com>
Message-ID: <SNT127-W651BB345720F0D2ADD03D6D5940@phx.gbl>


Avi,
Yes you are correct. Cdrs via xml_cdr. I have done this from httpd side already. I tried to create some method with process_cdr false but without any luck.
Thanks.
From: avi at avimarcus.net
Date: Tue, 18 Sep 2012 20:14:01 +0300
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to disable adding of rows with LOSE_RACE case ?

If you use xml_cdr or json_cdr that posts to a database, you can just drop those CDRs.
I wonder if you can use a hangup hook of a sort (before reporting!) to set process_cdr to false, or a_only...


-Avi


On Tue, Sep 18, 2012 at 5:11 PM, Yuriy Nasida <nasida at live.ru> wrote:






Hello guys!
I use many endpoints in my bridge appl. So in cdr I see one good row (for ext which picks up call) and  many rows with "LOSE_RACE" for others ext from bridge string. How can I disable adding of rows with LOSE_RACE ? 


Please advice.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



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

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

http://www.freeswitch.org





_________________________________________________________________________
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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FreeSWITCH-users mailing list
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From msc at freeswitch.org  Wed Sep 19 02:00:06 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 15:00:06 -0700
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <80eb858fe62119a6c53d369db8d61893.squirrel@fruiteater.riseup.net>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
	<5058E2C9.2080308@cupis.co.uk>
	<65d848bade245bbb25d895325f570d28.squirrel@fruiteater.riseup.net>
	<CAKzWOxX2BydfAaTL-D80vtRBgbF-zw+9sEYwRyg5m-+y8nbcLg@mail.gmail.com>
	<80eb858fe62119a6c53d369db8d61893.squirrel@fruiteater.riseup.net>
Message-ID: <CAKzWOxUBgx-m+o=o0D_EbORZn9_mKmiunz2MPLR1QHj8TPLWkA@mail.gmail.com>

We prefer not to store same info in multiple places unless there's no
elegant way around it. In this case you're fine just to put this tidbit
into both pages.
-MC

On Tue, Sep 18, 2012 at 2:46 PM, <georg at riseup.net> wrote:

> Hi,
>
> > Where on the wiki were you looking for this information? It sounds like
> it
> > is related to caller id handling. My guess is that the best place to put
> > it
> > would be:
> > http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
>
> Allright, will put it there.
> Is it undesired to store the same information twice? Because if not, I
> would also add it to the dialplan examples, maybe next to [1].
>
> Cheers,
> Georg
>
> [1]
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_3:_Stripping_leading_digits
>
> > if you put it inside of the <onlyinclude> tags then it will also display
> > on
> > the main channel variables page.
> >
> > -MC
> >
> > On Tue, Sep 18, 2012 at 2:20 PM, <georg at riseup.net> wrote:
> >
> >> Hi Paul,
> >>
> >> > Try this instead of the above:
> >> >
> >> > <action application="set"
> >> >
> >>
> data="effective_caller_id_number=${regex(${caller_id_number}|^49([0-9]{9})$|0%1)}"/>
> >> >
> >> > You maybe want to adjust the {9} to reflect the correct number of
> >> digits
> >> > in the national part of the caller ID.
> >>
> >> Thank you. I've found this aswell in the wiki 2 minutes after my post,
> >> and
> >> wanted to test first.
> >> Everything works as expected! :)
> >>
> >> Cheers,
> >> Georg
> >>
> >> P.S.: Before my post I googled aroung for sth. like two hours. Should I
> >> wikify this somewhere (if yes, where?) to make it easier to find?
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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 msc at freeswitch.org  Wed Sep 19 02:00:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Sep 2012 15:00:35 -0700
Subject: [Freeswitch-users] Strip digits from incoming caller id
In-Reply-To: <CALNZuwmptSiyvAnVBnJwOxRWpcOxfLD096wC7YzKtHPsL74-kQ@mail.gmail.com>
References: <45ece649c482ef0d4d260a5217a7cc6d.squirrel@fruiteater.riseup.net>
	<5058E2C9.2080308@cupis.co.uk>
	<65d848bade245bbb25d895325f570d28.squirrel@fruiteater.riseup.net>
	<CAKzWOxX2BydfAaTL-D80vtRBgbF-zw+9sEYwRyg5m-+y8nbcLg@mail.gmail.com>
	<CALNZuwmptSiyvAnVBnJwOxRWpcOxfLD096wC7YzKtHPsL74-kQ@mail.gmail.com>
Message-ID: <CAKzWOxUj_CWsMoJkfkR-s7+QV=Y3rZRmsJLPOS6widOfY2BWCg@mail.gmail.com>

On Tue, Sep 18, 2012 at 2:45 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Whoa. I did NOT know about this onlyinclude stuff.
> It looks like there's quite a few vars on channel vars page that aren't
> using an include like this.
>
> -Avi
>
>
This is a great janitorial project if anyone wants to help. It's very easy
to do - it's just tedious.
-MC
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From vipkilla at gmail.com  Wed Sep 19 02:48:53 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Tue, 18 Sep 2012 18:48:53 -0400
Subject: [Freeswitch-users] no audio from portaudio in conference
In-Reply-To: <CALn233oPfO5EfjTVVpoU2MGUvCgWzJZBG=mUWZhoR3-jumpKKg@mail.gmail.com>
References: <CAC-LwPPEw641dyzuWU6OwTQYTqfNfsoTGr9xfV-51EGydc1ZuQ@mail.gmail.com>
	<CALn233o_r-CTdMf0of-vwQ-+ro9aGf9jhCV+2rR+qWEru1J1MA@mail.gmail.com>
	<CAC-LwPM5ikXpyjP90Z_fXPqZoD321gqvM-+-3kCwSXyWkWesRA@mail.gmail.com>
	<CALn233oPfO5EfjTVVpoU2MGUvCgWzJZBG=mUWZhoR3-jumpKKg@mail.gmail.com>
Message-ID: <CAC-LwPOnTLBB1HuSHSW-kYMmSb2yR_aa+HQKrty43u=T7TYpnQ@mail.gmail.com>

>> On Mon, Sep 17, 2012 at 11:52 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
>>> FSClient uses portaudio at its core and is built against 1.2 it even
>>> does conferencing.   Does portaudio work for a normal call?   Is it
>>> just the mic input not working or is the speakerphone output also not
>>> working?
>>>
>>> ~Mitch


The speaker output does not work for this portaudio device either. Yet
it works in freeswitch-1.07


From moises.silva at gmail.com  Wed Sep 19 02:57:00 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 18 Sep 2012 18:57:00 -0400
Subject: [Freeswitch-users] freetdm and destination_number
In-Reply-To: <5036F6CE.3010207@synacktics.com>
References: <5036F6CE.3010207@synacktics.com>
Message-ID: <CAA4nhyCfZYnRDLMF-vGCur41sSDmb=5t3FM+=1f+hhLm95h_Ag@mail.gmail.com>

On Thu, Aug 23, 2012 at 11:36 PM, Tim Bock <jtbock at synacktics.com> wrote:

>
> 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.
>
>
The problem is that fxo lines do not receive a destination number, they
just "ring". You can set a fixed destination number though, just use the
"number => " configuration in freetdm.conf

ie

[span zt myspan]
number => 1234
fxo-channel => 1


*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>

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From dujinfang at gmail.com  Wed Sep 19 03:35:13 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 19 Sep 2012 07:35:13 +0800
Subject: [Freeswitch-users] mod_conference wrong codec for auto_outcall
 b-leg
In-Reply-To: <5058759C.9050506@sysvision.de>
References: <5058759C.9050506@sysvision.de>
Message-ID: <D7E2D92BA89A47BBA28012CE29BEBCF4@gmail.com>

maybe try to set http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix ?

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, September 18, 2012 at 9:22 PM, Oliver Saggau | Sysvision GmbH wrote:

> Hi everyone,
> I got a strange problem with mod_conference and auto_outcall taking the 
> wrong codec for b-leg. For what I can see from the log everything seems 
> fine. The a-leg calls with the following SDP:
> 
> 2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP:
> v=0
> o=- 3556961565 3556961565 IN IP4 10.20.30.125
> s=pjmedia
> t=0 0
> m=audio 4000 RTP/AVP 99 101
> c=IN IP4 10.20.30.125
> a=rtpmap:99 SILK/24000
> a=fmtp:99 useinbandfec=0
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtcp:4001 IN IP4 10.20.30.125
> 
> And the SILK/24000 codec is accepted according to the log file:
> 2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec 
> sofia/internal/1000004 at 10.20.30.240 (mailto:1000004 at 10.20.30.240) SILK/24000 20 ms 480 samples 40000 bits
> 2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111 
> sofia/internal/1000004 at 10.20.30.240 (mailto:1000004 at 10.20.30.240) Original read codec set to SILK:120
> 2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP 
> sofia/internal/1000004 at 10.20.30.240 (mailto:1000004 at 10.20.30.240):
> v=0
> o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240
> s=FreeSWITCH
> c=IN IP4 10.20.30.240
> t=0 0
> m=audio 21622 RTP/AVP 99 101
> a=rtpmap:99 SILK/24000
> a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> The conference get's created and activates the read/write codecs for the 
> first member:
> 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec 
> Activation Success L16 at 24000hz 1 channel 20ms
> 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec 
> Activation Success L16 at 48000hz 1 channel 20ms
> 
> After that he's trying to do the auto_outcall and that's where things 
> get weird:
> EXECUTE sofia/internal/1000004 at freeswitch.aws.hosts.corpex.de (mailto:1000004 at freeswitch.aws.hosts.corpex.de) 
> conference_set_auto_outcall(sofia/internal/sip:1000003 at 10.20.30.108 (mailto:1000003 at 10.20.30.108):63741;transport=UDP;ob)
> 2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel 
> sofia/internal/sip:1000003 at 10.20.30.108 (mailto:1000003 at 10.20.30.108):63741 
> [0236d005-1ff0-4f13-9370-bf7736087dd0]
> 2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP:
> v=0
> o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240
> s=FreeSWITCH
> c=IN IP4 10.20.30.240
> t=0 0
> m=audio 20318 RTP/AVP 98 0 8 101 13
> a=rtpmap:98 SILK/8000
> a=fmtp:98 useinbandfec=1; usedtx=0
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec 
> Activation Success L16 at 8000hz 1 channel 20ms
> 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec 
> Activation Success L16 at 48000hz 1 channel 20ms
> 
> Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is 
> configured like this:
> outbound_codec_prefs=silk at 24000,silk at 16000,silk at 8000,PCMU,PCMA,GSM
> 
> Thanks for your help,
> Oliver
> 
> _________________________________________________________________________
> 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 mitch.capper at gmail.com  Wed Sep 19 05:17:54 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 18 Sep 2012 18:17:54 -0700
Subject: [Freeswitch-users] no audio from portaudio in conference
In-Reply-To: <CAC-LwPOnTLBB1HuSHSW-kYMmSb2yR_aa+HQKrty43u=T7TYpnQ@mail.gmail.com>
References: <CAC-LwPPEw641dyzuWU6OwTQYTqfNfsoTGr9xfV-51EGydc1ZuQ@mail.gmail.com>
	<CALn233o_r-CTdMf0of-vwQ-+ro9aGf9jhCV+2rR+qWEru1J1MA@mail.gmail.com>
	<CAC-LwPM5ikXpyjP90Z_fXPqZoD321gqvM-+-3kCwSXyWkWesRA@mail.gmail.com>
	<CALn233oPfO5EfjTVVpoU2MGUvCgWzJZBG=mUWZhoR3-jumpKKg@mail.gmail.com>
	<CAC-LwPOnTLBB1HuSHSW-kYMmSb2yR_aa+HQKrty43u=T7TYpnQ@mail.gmail.com>
Message-ID: <CALn233ouQNK-r=-YoKfmYEVn6A_ipyPyxmNO0F=cOOxy-Z-bdg@mail.gmail.com>

Does it work on a normal call? What does pa devlist show? Is there
anything in the console logs when set to debug (try fs_logger.pl to
pastebin the logs).

~Mitch

On Tue, Sep 18, 2012 at 3:48 PM, Vik Killa <vipkilla at gmail.com> wrote:
>>> On Mon, Sep 17, 2012 at 11:52 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
>>>> FSClient uses portaudio at its core and is built against 1.2 it even
>>>> does conferencing.   Does portaudio work for a normal call?   Is it
>>>> just the mic input not working or is the speakerphone output also not
>>>> working?
>>>>
>>>> ~Mitch
>
>
> The speaker output does not work for this portaudio device either. Yet
> it works in freeswitch-1.07
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jack at livecall.com  Wed Sep 19 06:44:40 2012
From: jack at livecall.com (Jack)
Date: Tue, 18 Sep 2012 19:44:40 -0700
Subject: [Freeswitch-users] is the example flex client can't receive a
 call?
In-Reply-To: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
Message-ID: <50593198.5040801@livecall.com>

I haven't tried it but this was on the mod_rtmp wiki for bridging.

This can be used to bridge a call to the user:

<action application="bridge" data="${rtmp_contact($${rtmp_profile}/${dialed_ext}@$${domain})}"/>


On 9/18/2012 10:45 AM, siklub wrote:
> Hi All,
> I just try the example flex client.
> It can make a call to 888 at conference.freeswitch.org
> But it can't receive a call.
> Some error message, maybe:
> switch_ivr_originate.c:2591 Cannot create outgoing channel of type
> [error] cause: [USER_NOT_REGISTERED]
> mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
> ....
> rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
>
> Is the example can receive a call? (it is only example, maybe it is incomplete).
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From klubvps at gmail.com  Wed Sep 19 06:53:13 2012
From: klubvps at gmail.com (siklub)
Date: Wed, 19 Sep 2012 09:53:13 +0700
Subject: [Freeswitch-users] is the example flex client can't receive a
	call?
In-Reply-To: <50593198.5040801@livecall.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
	<50593198.5040801@livecall.com>
Message-ID: <CAH3d9AmU+L=6O0iNQ+MjVx1eyou5UBPykDtvHBPQx4QgtmJFjA@mail.gmail.com>

On Wed, Sep 19, 2012 at 9:44 AM, Jack <jack at livecall.com> wrote:
> I haven't tried it but this was on the mod_rtmp wiki for bridging.
>
> This can be used to bridge a call to the user:
>
> <action application="bridge" data="${rtmp_contact($${rtmp_profile}/${dialed_ext}@$${domain})}"/>

Thanks Jack,
Actually, i'm very new to freeswitch. I still learn to add a user when
reading your email.

Where do i must put that line?

How about the other line mentioned in the wiki?
rtmp_contact profile/user at domain[/[!]nickname]

Do you have some material that recommended to understand this?
I thought that a user connecting from xlite, ekiga, or sflphone is the
same as a user connecting from flex client.

>
>
> On 9/18/2012 10:45 AM, siklub wrote:
>> Hi All,
>> I just try the example flex client.
>> It can make a call to 888 at conference.freeswitch.org
>> But it can't receive a call.
>> Some error message, maybe:
>> switch_ivr_originate.c:2591 Cannot create outgoing channel of type
>> [error] cause: [USER_NOT_REGISTERED]
>> mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
>> ....
>> rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
>>
>> Is the example can receive a call? (it is only example, maybe it is incomplete).
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep 19 08:32:15 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 00:32:15 -0400
Subject: [Freeswitch-users] How to force FS to execute?
Message-ID: <17B158D4-648A-4D19-BF4D-48E2A861479D@kavun.ch>

Hi all,

My example below cannot work because FS does not store the value of ${pin} until the call goes into execution state. While keeping it simple, what would be your take on this? Is there a way to instruct FS to start executing at some point and discovering the dialplan along the way?

<extension name="pin test">
<condition field="destination_number" expression="^111$" break="never">
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*" inline="true" />
</condition>
<condition field="${pin}" expression="^1234$">
<action application="transfer" data ="3000 XML local" />
</condition>
<condition field="${pin}" expression="^\d{4}$">
<action application="hangup" />
</condition>
</extension>

Thanks and all the best, as always.
Emrah

From lists at kavun.ch  Wed Sep 19 08:36:38 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 00:36:38 -0400
Subject: [Freeswitch-users] You don't need to save a greeting to
	override your existing one in Voicemail
In-Reply-To: <CAKzWOxUrcurCUfgrVnyp2jWqv6OJQfc=hRnXX7RKayLD4=cH5g@mail.gmail.com>
References: <F07A307C-BE0A-43C7-8C88-0B7C411C7D8E@kavun.ch>
	<CAKzWOxUrcurCUfgrVnyp2jWqv6OJQfc=hRnXX7RKayLD4=cH5g@mail.gmail.com>
Message-ID: <1FBE5249-F5EF-43A5-ABB4-F985359A4185@kavun.ch>

MC, that is the solution I use. I am not too bothered by this issue, just wanted to flag it up as a nice to have for future upgrades. 
As it is now, the "save option" is basically a "go to advanced option menu".

E
On Sep 17, 2012, at 11:35 AM, Michael Collins <msc at freeswitch.org> wrote:

> Emrah,
> 
> I think this is doable with some programming in mod_voicemail.c although I don't believe it's a high priority at the moment. However, you can manually emulate this behavior by recording a new greeting number that is different from your current, active greeting. For example, if your current greeting is 1, then record greeting 2, make sure it's satisfactory, then set 2 as your active greeting. It is basically an extra step, but at least it doesn't require any coding changes and it will get the job done.
> 
> -MC
> 
> On Fri, Sep 14, 2012 at 2:30 PM, Emrah <lists at kavun.ch> wrote:
> It used to be the case with Asterisk as well, now it has changed.
> Basically, if you have a greeting in place and want to record a new one? And you suddenly cough in the middle of your recording? Than get interrupted by your boss right after you've pressed #? You just hang up the phone and think that your existing greeting hasn't been altered because you didn't press 2 to save your re-recorded greeting?
> Well that's wrong.
> 
> Your existing greeting gets overridden as soon as you start the recording. Even if you hang up the phone in the middle of your message, it'll still be recorded and played as your greeting.
> 
> I don't think that this calls for a workaround or more studying of the app. I am using the default settings on this and think that it can be improved, so that you can review your greeting before committing it with a save action.
> Being able to listen to your existing greeting before re-recording it wouldn't be a bad add-on either.
> 
> What is your take on this?
> 
> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep 19 08:42:27 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 00:42:27 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <17B158D4-648A-4D19-BF4D-48E2A861479D@kavun.ch>
References: <17B158D4-648A-4D19-BF4D-48E2A861479D@kavun.ch>
Message-ID: <6AB70119-ABAC-420A-A8D1-E8976E9C2020@kavun.ch>

Please bypass the inline="true"? A fantasy of mine.
On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:

> Hi all,
> 
> My example below cannot work because FS does not store the value of ${pin} until the call goes into execution state. While keeping it simple, what would be your take on this? Is there a way to instruct FS to start executing at some point and discovering the dialplan along the way?
> 
> <extension name="pin test">
> <condition field="destination_number" expression="^111$" break="never">
> <action application="answer"/>
> <action application="sleep" data="1000"/>
> <action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*" inline="true" />
> </condition>
> <condition field="${pin}" expression="^1234$">
> <action application="transfer" data ="3000 XML local" />
> </condition>
> <condition field="${pin}" expression="^\d{4}$">
> <action application="hangup" />
> </condition>
> </extension>
> 
> Thanks and all the best, as always.
> Emrah



From krice at freeswitch.org  Wed Sep 19 08:50:17 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 18 Sep 2012 23:50:17 -0500
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <6AB70119-ABAC-420A-A8D1-E8976E9C2020@kavun.ch>
Message-ID: <CC7EB939.4D3D7%krice@freeswitch.org>

This is easy to handle... Do it as 2 different extensions and after you
collect the PIN, use the transfer application to transfer the call to a pin
validating extension... Problem fixed with a very simple solution

On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:

> Please bypass the inline="true"? A fantasy of mine.
> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> 
>> Hi all,
>> 
>> My example below cannot work because FS does not store the value of ${pin}
>> until the call goes into execution state. While keeping it simple, what would
>> be your take on this? Is there a way to instruct FS to start executing at
>> some point and discovering the dialplan along the way?
>> 
>> <extension name="pin test">
>> <condition field="destination_number" expression="^111$" break="never">
>> <action application="answer"/>
>> <action application="sleep" data="1000"/>
>> <action application="read" data="4 4
>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"
>> inline="true" />
>> </condition>
>> <condition field="${pin}" expression="^1234$">
>> <action application="transfer" data ="3000 XML local" />
>> </condition>
>> <condition field="${pin}" expression="^\d{4}$">
>> <action application="hangup" />
>> </condition>
>> </extension>
>> 
>> Thanks and all the best, as always.
>> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From klubvps at gmail.com  Wed Sep 19 09:01:29 2012
From: klubvps at gmail.com (siklub)
Date: Wed, 19 Sep 2012 12:01:29 +0700
Subject: [Freeswitch-users] is the example flex client can't receive a
	call?
In-Reply-To: <CAH3d9AmU+L=6O0iNQ+MjVx1eyou5UBPykDtvHBPQx4QgtmJFjA@mail.gmail.com>
References: <CAH3d9AnHkLC_O27Qg-5BEvoCiy93WDF=ooP0LKQYbJM2mS5VOA@mail.gmail.com>
	<50593198.5040801@livecall.com>
	<CAH3d9AmU+L=6O0iNQ+MjVx1eyou5UBPykDtvHBPQx4QgtmJFjA@mail.gmail.com>
Message-ID: <CAH3d9Aks9tDu6pTFU+4k59-MQMs5=ByhLzX+V2Pus=DjWVKgXA@mail.gmail.com>

Hi,
I have added
<action application="bridge"
data="${rtmp_contact($${rtmp_profile}/${dialed_ext}@$${domain})}"/>
in conf/dialplan/default.xml in Local_Extension

and set <param name="context" value="default" /> in rtmp.conf.xml

Now, the error message says:

2012-09-19 11:52:33.848536 [ERR] switch_core_session.c:408 Could not
locate channel type Usage: rtmp_contact profile
2012-09-19 11:52:33.848536 [NOTICE] switch_ivr_originate.c:2591 Cannot
create outgoing channel of type [Usage: rtmp_contact profile] cause:
[CHAN_NOT_IMPLEMENTED]
2012-09-19 11:52:33.848536 [INFO] mod_dptools.c:3027 Originate Failed.
 Cause: CHAN_NOT_IMPLEMENTED


Any clue?

Thanks

On Wed, Sep 19, 2012 at 9:53 AM, siklub <klubvps at gmail.com> wrote:
> On Wed, Sep 19, 2012 at 9:44 AM, Jack <jack at livecall.com> wrote:
>> I haven't tried it but this was on the mod_rtmp wiki for bridging.
>>
>> This can be used to bridge a call to the user:
>>
>> <action application="bridge" data="${rtmp_contact($${rtmp_profile}/${dialed_ext}@$${domain})}"/>
>
> Thanks Jack,
> Actually, i'm very new to freeswitch. I still learn to add a user when
> reading your email.
>
> Where do i must put that line?
>
> How about the other line mentioned in the wiki?
> rtmp_contact profile/user at domain[/[!]nickname]
>
> Do you have some material that recommended to understand this?
> I thought that a user connecting from xlite, ekiga, or sflphone is the
> same as a user connecting from flex client.
>
>>
>>
>> On 9/18/2012 10:45 AM, siklub wrote:
>>> Hi All,
>>> I just try the example flex client.
>>> It can make a call to 888 at conference.freeswitch.org
>>> But it can't receive a call.
>>> Some error message, maybe:
>>> switch_ivr_originate.c:2591 Cannot create outgoing channel of type
>>> [error] cause: [USER_NOT_REGISTERED]
>>> mod_dptools.c:3027 Originate Failed.  Cause: USER_NOT_REGISTERED
>>> ....
>>> rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3)
>>>
>>> Is the example can receive a call? (it is only example, maybe it is incomplete).
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From peetzer at gmail.com  Wed Sep 19 10:01:23 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Wed, 19 Sep 2012 08:01:23 +0200
Subject: [Freeswitch-users] ESL API command and ringback
Message-ID: <CACSMF4wT1qpB1R-exHx1QPmkHvOy+CP1+kTLQ=Aa3JKeRke+Bw@mail.gmail.com>

Hi again [?],

Simple question,

In the dialplan there is a Local_Extension to handle calls.

In this example, you have an action;

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

As api command (from ESL java outbound example) I put this like;


String result = doAction("global_getvar", "us-ring");// using CLI
(inbound), result  *%(2000,4000,440,480)* as defined in vars.xml

SendMsg sendMsg = *new* SendMsg();

sendMsg.addCallCommand("execute");

sendMsg.addExecuteAppName("set");

sendMsg.addExecuteAppArg("ringback=" + result);

but I don't hear sound. I don't know if the string argument "${us-string}",
can be parsed directly.

I tried everything but no sound.

Only thing that I got working is by using "execute" and "ring_ready"

What I'm doing wrong with ringback argument?

Kind regards,

Peter
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From avi at avimarcus.net  Wed Sep 19 11:49:59 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Sep 2012 10:49:59 +0300
Subject: [Freeswitch-users] You don't need to save a greeting to
 override your existing one in Voicemail
In-Reply-To: <1FBE5249-F5EF-43A5-ABB4-F985359A4185@kavun.ch>
References: <F07A307C-BE0A-43C7-8C88-0B7C411C7D8E@kavun.ch>
	<CAKzWOxUrcurCUfgrVnyp2jWqv6OJQfc=hRnXX7RKayLD4=cH5g@mail.gmail.com>
	<1FBE5249-F5EF-43A5-ABB4-F985359A4185@kavun.ch>
Message-ID: <CALNZuwnfkokrvAV+n9ixi9iOAiE65q6vt9TFnhRegcYLkW0KRQ@mail.gmail.com>

To flag it, open a Jira -- http://jira.freeswitch.org

-Avi


On Wed, Sep 19, 2012 at 7:36 AM, Emrah <lists at kavun.ch> wrote:

> MC, that is the solution I use. I am not too bothered by this issue, just
> wanted to flag it up as a nice to have for future upgrades.
> As it is now, the "save option" is basically a "go to advanced option
> menu".
>
> E
> On Sep 17, 2012, at 11:35 AM, Michael Collins <msc at freeswitch.org> wrote:
>
> > Emrah,
> >
> > I think this is doable with some programming in mod_voicemail.c although
> I don't believe it's a high priority at the moment. However, you can
> manually emulate this behavior by recording a new greeting number that is
> different from your current, active greeting. For example, if your current
> greeting is 1, then record greeting 2, make sure it's satisfactory, then
> set 2 as your active greeting. It is basically an extra step, but at least
> it doesn't require any coding changes and it will get the job done.
> >
> > -MC
> >
> > On Fri, Sep 14, 2012 at 2:30 PM, Emrah <lists at kavun.ch> wrote:
> > It used to be the case with Asterisk as well, now it has changed.
> > Basically, if you have a greeting in place and want to record a new one?
> And you suddenly cough in the middle of your recording? Than get
> interrupted by your boss right after you've pressed #? You just hang up the
> phone and think that your existing greeting hasn't been altered because you
> didn't press 2 to save your re-recorded greeting?
> > Well that's wrong.
> >
> > Your existing greeting gets overridden as soon as you start the
> recording. Even if you hang up the phone in the middle of your message,
> it'll still be recorded and played as your greeting.
> >
> > I don't think that this calls for a workaround or more studying of the
> app. I am using the default settings on this and think that it can be
> improved, so that you can review your greeting before committing it with a
> save action.
> > Being able to listen to your existing greeting before re-recording it
> wouldn't be a bad add-on either.
> >
> > What is your take on this?
> >
> > 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
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From royj at yandex.ru  Wed Sep 19 12:40:42 2012
From: royj at yandex.ru (royj)
Date: Wed, 19 Sep 2012 12:40:42 +0400
Subject: [Freeswitch-users] t38 issue
In-Reply-To: <mailman.18724.1347388597.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.18724.1347388597.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <20120919124042.c067bbb3cd9592c47edb500d@yandex.ru>


Thank you for all your assistance

Yesterday FreeSWITCH have been updated to Version 1.3.0+git~20120918T210019Z~d80e91d248 (1.3.0; git at commit d80e91d248 on Tue, 18 Sep 2012 21:00:19 Z) and no changes.

Maybe there is some specific settings, that I miss?

Does anybody has been solving this mission?


On 9/11/12 13:36 AM, "Ken Rice" <krice at freeswitch.org> wrote:
> 
> There have been updates to t.38 since 1.2.0-rc2...  You should either be on
> head or v1.2.stable branch
> 
> 
> 
> 
> On 9/11/12 10:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
> 
> > I think the guys have made some improvements since 1.2.0-rc2. Any chance you
> > can get at least up to 1.2.1-stable and retest?
> > -MC
> > 
> > On Tue, Sep 11, 2012 at 1:48 AM, royj <royj at yandex.ru> wrote:
> >> 
> >> Hi all
> >> 
> >> There is some strange issue with t38-passthru (FreeSWITCH Version 1.2.0-rc2).
> >> Task - t38-passthru
> >> spa112(ip 1.1.1.1) <--(profile local)--> FreeSWITCH(ip 2.2.2.2)<--(profile
> >> extrernal)--> Mediant 2000(ip 3.3.3.3); fax from spa112 to Mediant 2000.
> >> In both profiles:
> >> <param name="t38-passthru" value="true"/>
> >> <param name="inbound-proxy-media" value="true"/>
> >> <param name="inbound-late-negotiation" value="true"/>
> >> 
> >> Here is the sip-trace of the call with fax -
> >> http://pastebin.freeswitch.org/19866
> >> 
> >> It is seen that t38 re-INVITE from spa112 (profile local) passed on and back
> >> with "200 OK", but t38 from Mediant 2000 (profile external) is not (in
> >> response to re-INVITE t38 (SDP, m=image) - "200 OK" with SDP m=audio). After
> >> agreement the SDP, the data (v21-preamble) is not passed by freeswitch
> >> between spa112 and Mediant 2000, and one of the party ends conversation with
> >> message "no-signal" and then "by" .
> >> 
> >> Wiki says ( 
> >> http://wiki.freeswitch.org/wiki/Proxy_Media#Why_you_almost_certainly_don.27t_
> >> want_to_use_it ) that module mod_spandsp is required (if I understand
> >> correctly), but "load mod_spandsp" does not change anything.
> >> 
> >> I would appreciate your help
> >> 
> >> --
> >> Regards
> >> royj
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> 
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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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> 
> ------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep 19 15:10:33 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 07:10:33 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <CC7EB939.4D3D7%krice@freeswitch.org>
References: <CC7EB939.4D3D7%krice@freeswitch.org>
Message-ID: <72930F09-E0E4-4EC4-AEC5-6ECD0AC14E66@kavun.ch>

Easy indeed, but I wanted to avoid it thinking there was some other alternative. 
Thanks for your quick response, will use it.

Cheers,
Emrah
On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:

> This is easy to handle... Do it as 2 different extensions and after you
> collect the PIN, use the transfer application to transfer the call to a pin
> validating extension... Problem fixed with a very simple solution
> 
> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> 
>> Please bypass the inline="true"? A fantasy of mine.
>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>> 
>>> Hi all,
>>> 
>>> My example below cannot work because FS does not store the value of ${pin}
>>> until the call goes into execution state. While keeping it simple, what would
>>> be your take on this? Is there a way to instruct FS to start executing at
>>> some point and discovering the dialplan along the way?
>>> 
>>> <extension name="pin test">
>>> <condition field="destination_number" expression="^111$" break="never">
>>> <action application="answer"/>
>>> <action application="sleep" data="1000"/>
>>> <action application="read" data="4 4
>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"
>>> inline="true" />
>>> </condition>
>>> <condition field="${pin}" expression="^1234$">
>>> <action application="transfer" data ="3000 XML local" />
>>> </condition>
>>> <condition field="${pin}" expression="^\d{4}$">
>>> <action application="hangup" />
>>> </condition>
>>> </extension>
>>> 
>>> Thanks and all the best, as always.
>>> 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
> 
> -- 
> 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



From lists at kavun.ch  Wed Sep 19 16:25:53 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 08:25:53 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <72930F09-E0E4-4EC4-AEC5-6ECD0AC14E66@kavun.ch>
References: <CC7EB939.4D3D7%krice@freeswitch.org>
	<72930F09-E0E4-4EC4-AEC5-6ECD0AC14E66@kavun.ch>
Message-ID: <B9B36BB1-1067-4043-8A39-625EE4359261@kavun.ch>

Is there a way to address an extension without assigning it to a destination_number? I don't want the extension to be reachable directly and would gladly avoid creating a dedicated context.
Can I transfer a call to an extension that doesn't have a destination_number?

Best,
Emrah
On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:

> Easy indeed, but I wanted to avoid it thinking there was some other alternative. 
> Thanks for your quick response, will use it.
> 
> Cheers,
> Emrah
> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
> 
>> This is easy to handle... Do it as 2 different extensions and after you
>> collect the PIN, use the transfer application to transfer the call to a pin
>> validating extension... Problem fixed with a very simple solution
>> 
>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>> 
>>> Please bypass the inline="true"? A fantasy of mine.
>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>>> 
>>>> Hi all,
>>>> 
>>>> My example below cannot work because FS does not store the value of ${pin}
>>>> until the call goes into execution state. While keeping it simple, what would
>>>> be your take on this? Is there a way to instruct FS to start executing at
>>>> some point and discovering the dialplan along the way?
>>>> 
>>>> <extension name="pin test">
>>>> <condition field="destination_number" expression="^111$" break="never">
>>>> <action application="answer"/>
>>>> <action application="sleep" data="1000"/>
>>>> <action application="read" data="4 4
>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"
>>>> inline="true" />
>>>> </condition>
>>>> <condition field="${pin}" expression="^1234$">
>>>> <action application="transfer" data ="3000 XML local" />
>>>> </condition>
>>>> <condition field="${pin}" expression="^\d{4}$">
>>>> <action application="hangup" />
>>>> </condition>
>>>> </extension>
>>>> 
>>>> Thanks and all the best, as always.
>>>> 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
>> 
>> -- 
>> 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
> 



From avi at avimarcus.net  Wed Sep 19 16:36:52 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Sep 2012 15:36:52 +0300
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <B9B36BB1-1067-4043-8A39-625EE4359261@kavun.ch>
References: <CC7EB939.4D3D7%krice@freeswitch.org>
	<72930F09-E0E4-4EC4-AEC5-6ECD0AC14E66@kavun.ch>
	<B9B36BB1-1067-4043-8A39-625EE4359261@kavun.ch>
Message-ID: <CALNZuwnMV=wMt0eNvPmj6VMLnM3OL-C+xK5Ayf=8df+yY19noA@mail.gmail.com>

You can create an extension that triggers based on whatever criteria you
want, e.g. any variable.

e.g. <condition field="${my_custom_handling}" expression="^done-pin$"
break="never">

-Avi


On Wed, Sep 19, 2012 at 3:25 PM, Emrah <lists at kavun.ch> wrote:

> Is there a way to address an extension without assigning it to a
> destination_number? I don't want the extension to be reachable directly and
> would gladly avoid creating a dedicated context.
> Can I transfer a call to an extension that doesn't have a
> destination_number?
>
> Best,
> Emrah
> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
>
> > Easy indeed, but I wanted to avoid it thinking there was some other
> alternative.
> > Thanks for your quick response, will use it.
> >
> > Cheers,
> > Emrah
> > On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
> >
> >> This is easy to handle... Do it as 2 different extensions and after you
> >> collect the PIN, use the transfer application to transfer the call to a
> pin
> >> validating extension... Problem fixed with a very simple solution
> >>
> >> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> >>
> >>> Please bypass the inline="true"? A fantasy of mine.
> >>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> >>>
> >>>> Hi all,
> >>>>
> >>>> My example below cannot work because FS does not store the value of
> ${pin}
> >>>> until the call goes into execution state. While keeping it simple,
> what would
> >>>> be your take on this? Is there a way to instruct FS to start
> executing at
> >>>> some point and discovering the dialplan along the way?
> >>>>
> >>>> <extension name="pin test">
> >>>> <condition field="destination_number" expression="^111$"
> break="never">
> >>>> <action application="answer"/>
> >>>> <action application="sleep" data="1000"/>
> >>>> <action application="read" data="4 4
> >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin
> 10000 #,*"
> >>>> inline="true" />
> >>>> </condition>
> >>>> <condition field="${pin}" expression="^1234$">
> >>>> <action application="transfer" data ="3000 XML local" />
> >>>> </condition>
> >>>> <condition field="${pin}" expression="^\d{4}$">
> >>>> <action application="hangup" />
> >>>> </condition>
> >>>> </extension>
> >>>>
> >>>> Thanks and all the best, as always.
> >>>> 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
> >>
> >> --
> >> 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
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Sep 19 17:11:01 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 09:11:01 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <CALNZuwnMV=wMt0eNvPmj6VMLnM3OL-C+xK5Ayf=8df+yY19noA@mail.gmail.com>
References: <CC7EB939.4D3D7%krice@freeswitch.org>
	<72930F09-E0E4-4EC4-AEC5-6ECD0AC14E66@kavun.ch>
	<B9B36BB1-1067-4043-8A39-625EE4359261@kavun.ch>
	<CALNZuwnMV=wMt0eNvPmj6VMLnM3OL-C+xK5Ayf=8df+yY19noA@mail.gmail.com>
Message-ID: <64D6D500-7F46-4302-90AE-5CA5C3D48309@kavun.ch>

Thanks Avi, I get that but how do you then transfer to that extension?
On Sep 19, 2012, at 8:36 AM, Avi Marcus <avi at avimarcus.net> wrote:

> You can create an extension that triggers based on whatever criteria you want, e.g. any variable.
> 
> e.g. <condition field="${my_custom_handling}" expression="^done-pin$" break="never">
> 
> -Avi
> 
> 
> On Wed, Sep 19, 2012 at 3:25 PM, Emrah <lists at kavun.ch> wrote:
> Is there a way to address an extension without assigning it to a destination_number? I don't want the extension to be reachable directly and would gladly avoid creating a dedicated context.
> Can I transfer a call to an extension that doesn't have a destination_number?
> 
> Best,
> Emrah
> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
> 
> > Easy indeed, but I wanted to avoid it thinking there was some other alternative.
> > Thanks for your quick response, will use it.
> >
> > Cheers,
> > Emrah
> > On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
> >
> >> This is easy to handle... Do it as 2 different extensions and after you
> >> collect the PIN, use the transfer application to transfer the call to a pin
> >> validating extension... Problem fixed with a very simple solution
> >>
> >> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> >>
> >>> Please bypass the inline="true"? A fantasy of mine.
> >>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> >>>
> >>>> Hi all,
> >>>>
> >>>> My example below cannot work because FS does not store the value of ${pin}
> >>>> until the call goes into execution state. While keeping it simple, what would
> >>>> be your take on this? Is there a way to instruct FS to start executing at
> >>>> some point and discovering the dialplan along the way?
> >>>>
> >>>> <extension name="pin test">
> >>>> <condition field="destination_number" expression="^111$" break="never">
> >>>> <action application="answer"/>
> >>>> <action application="sleep" data="1000"/>
> >>>> <action application="read" data="4 4
> >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"
> >>>> inline="true" />
> >>>> </condition>
> >>>> <condition field="${pin}" expression="^1234$">
> >>>> <action application="transfer" data ="3000 XML local" />
> >>>> </condition>
> >>>> <condition field="${pin}" expression="^\d{4}$">
> >>>> <action application="hangup" />
> >>>> </condition>
> >>>> </extension>
> >>>>
> >>>> Thanks and all the best, as always.
> >>>> 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
> >>
> >> --
> >> 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
> >
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From oliver.saggau at sysvision.de  Wed Sep 19 12:36:33 2012
From: oliver.saggau at sysvision.de (Oliver Saggau | Sysvision GmbH)
Date: Wed, 19 Sep 2012 10:36:33 +0200
Subject: [Freeswitch-users] mod_conference wrong codec for auto_outcall
 b-leg
In-Reply-To: <D7E2D92BA89A47BBA28012CE29BEBCF4@gmail.com>
References: <5058759C.9050506@sysvision.de>
	<D7E2D92BA89A47BBA28012CE29BEBCF4@gmail.com>
Message-ID: <50598411.90206@sysvision.de>

Do you have some options/variables in mind I should try using 
auto_outcall_prefix? I tried setting codec_string=silk at 24000 and 
absolute_codec_string=silk at 24000, but still FS offers only SILK/8000 to 
the b-leg.

Am 19.09.2012 01:35, schrieb Seven Du:
> maybe try to set 
> http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix ?
>
> -- 
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Tuesday, September 18, 2012 at 9:22 PM, Oliver Saggau | Sysvision 
> GmbH wrote:
>
>> Hi everyone,
>> I got a strange problem with mod_conference and auto_outcall taking the
>> wrong codec for b-leg. For what I can see from the log everything seems
>> fine. The a-leg calls with the following SDP:
>>
>> 2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP:
>> v=0
>> o=- 3556961565 3556961565 IN IP4 10.20.30.125
>> s=pjmedia
>> t=0 0
>> m=audio 4000 RTP/AVP 99 101
>> c=IN IP4 10.20.30.125
>> a=rtpmap:99 SILK/24000
>> a=fmtp:99 useinbandfec=0
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtcp:4001 IN IP4 10.20.30.125
>>
>> And the SILK/24000 codec is accepted according to the log file:
>> 2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec
>> sofia/internal/1000004 at 10.20.30.240 <mailto:1000004 at 10.20.30.240> 
>> SILK/24000 20 ms 480 samples 40000 bits
>> 2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111
>> sofia/internal/1000004 at 10.20.30.240 <mailto:1000004 at 10.20.30.240> 
>> Original read codec set to SILK:120
>> 2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP
>> sofia/internal/1000004 at 10.20.30.240 <mailto:1000004 at 10.20.30.240>:
>> v=0
>> o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240
>> s=FreeSWITCH
>> c=IN IP4 10.20.30.240
>> t=0 0
>> m=audio 21622 RTP/AVP 99 101
>> a=rtpmap:99 SILK/24000
>> a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> The conference get's created and activates the read/write codecs for the
>> first member:
>> 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec
>> Activation Success L16 at 24000hz 1 channel 20ms
>> 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec
>> Activation Success L16 at 48000hz 1 channel 20ms
>>
>> After that he's trying to do the auto_outcall and that's where things
>> get weird:
>> EXECUTE sofia/internal/1000004 at freeswitch.aws.hosts.corpex.de 
>> <mailto:1000004 at freeswitch.aws.hosts.corpex.de>
>> conference_set_auto_outcall(sofia/internal/sip:1000003 at 10.20.30.108 
>> <mailto:1000003 at 10.20.30.108>:63741;transport=UDP;ob)
>> 2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel
>> sofia/internal/sip:1000003 at 10.20.30.108 
>> <mailto:1000003 at 10.20.30.108>:63741
>> [0236d005-1ff0-4f13-9370-bf7736087dd0]
>> 2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP:
>> v=0
>> o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240
>> s=FreeSWITCH
>> c=IN IP4 10.20.30.240
>> t=0 0
>> m=audio 20318 RTP/AVP 98 0 8 101 13
>> a=rtpmap:98 SILK/8000
>> a=fmtp:98 useinbandfec=1; usedtx=0
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>> 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec
>> Activation Success L16 at 8000hz 1 channel 20ms
>> 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec
>> Activation Success L16 at 48000hz 1 channel 20ms
>>
>> Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is
>> configured like this:
>> outbound_codec_prefs=silk at 24000,silk at 16000,silk at 8000,PCMU,PCMA,GSM
>>
>> Thanks for your help,
>> Oliver
>>
>> _________________________________________________________________________
>> 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 avi at avimarcus.net  Wed Sep 19 17:25:28 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Sep 2012 16:25:28 +0300
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <64D6D500-7F46-4302-90AE-5CA5C3D48309@kavun.ch>
References: <CC7EB939.4D3D7%krice@freeswitch.org>
	<72930F09-E0E4-4EC4-AEC5-6ECD0AC14E66@kavun.ch>
	<B9B36BB1-1067-4043-8A39-625EE4359261@kavun.ch>
	<CALNZuwnMV=wMt0eNvPmj6VMLnM3OL-C+xK5Ayf=8df+yY19noA@mail.gmail.com>
	<64D6D500-7F46-4302-90AE-5CA5C3D48309@kavun.ch>
Message-ID: <CALNZuwkz=WHj0zC_eP_WfzAbLhUbuS9Q=-RBLyJOAS8HQrTcdQ@mail.gmail.com>

Oh, right.. transfer sets the destination number... You can set an
alphabetic prefix, e.g. "special_pin" destination, so normal users
can't accidentally dial the extension.

-Avi



On Wed, Sep 19, 2012 at 4:11 PM, Emrah <lists at kavun.ch> wrote:

> Thanks Avi, I get that but how do you then transfer to that extension?
> On Sep 19, 2012, at 8:36 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
> > You can create an extension that triggers based on whatever criteria you
> want, e.g. any variable.
> >
> > e.g. <condition field="${my_custom_handling}" expression="^done-pin$"
> break="never">
> >
> > -Avi
> >
> >
> > On Wed, Sep 19, 2012 at 3:25 PM, Emrah <lists at kavun.ch> wrote:
> > Is there a way to address an extension without assigning it to a
> destination_number? I don't want the extension to be reachable directly and
> would gladly avoid creating a dedicated context.
> > Can I transfer a call to an extension that doesn't have a
> destination_number?
> >
> > Best,
> > Emrah
> > On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
> >
> > > Easy indeed, but I wanted to avoid it thinking there was some other
> alternative.
> > > Thanks for your quick response, will use it.
> > >
> > > Cheers,
> > > Emrah
> > > On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
> > >
> > >> This is easy to handle... Do it as 2 different extensions and after
> you
> > >> collect the PIN, use the transfer application to transfer the call to
> a pin
> > >> validating extension... Problem fixed with a very simple solution
> > >>
> > >> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> > >>
> > >>> Please bypass the inline="true"? A fantasy of mine.
> > >>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> > >>>
> > >>>> Hi all,
> > >>>>
> > >>>> My example below cannot work because FS does not store the value of
> ${pin}
> > >>>> until the call goes into execution state. While keeping it simple,
> what would
> > >>>> be your take on this? Is there a way to instruct FS to start
> executing at
> > >>>> some point and discovering the dialplan along the way?
> > >>>>
> > >>>> <extension name="pin test">
> > >>>> <condition field="destination_number" expression="^111$"
> break="never">
> > >>>> <action application="answer"/>
> > >>>> <action application="sleep" data="1000"/>
> > >>>> <action application="read" data="4 4
> > >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin
> 10000 #,*"
> > >>>> inline="true" />
> > >>>> </condition>
> > >>>> <condition field="${pin}" expression="^1234$">
> > >>>> <action application="transfer" data ="3000 XML local" />
> > >>>> </condition>
> > >>>> <condition field="${pin}" expression="^\d{4}$">
> > >>>> <action application="hangup" />
> > >>>> </condition>
> > >>>> </extension>
> > >>>>
> > >>>> Thanks and all the best, as always.
> > >>>> 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
> > >>
> > >> --
> > >> 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
> > >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yehavi.bourvine at gmail.com  Wed Sep 19 18:31:30 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 19 Sep 2012 17:31:30 +0300
Subject: [Freeswitch-users] limit_usage change in recent versions?
Message-ID: <CAAtWKeGStyXKm4+o9FKJFAGNski=06RBjGSR54No+SQHYk_SFQ@mail.gmail.com>

Hello,

  I am using limit_usage in the following format:

<action application="bridge" data="${cond(${limit_usage(db, max_calls
${destination_number})} > ${MaxLimit} ?  error/user_busy :
${destination_mumber})}" />

It worked at older versions (my local reference is 1.0.6), but it doesn't
work in newer ones. It always returns zero.

All I want is to know whether the extension already has an active call to
decide what to do in case it is busy. What is the correct way of calling it?

                    Thanks, __Yehavi:
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From krice at freeswitch.org  Wed Sep 19 18:38:52 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Sep 2012 09:38:52 -0500
Subject: [Freeswitch-users] limit_usage change in recent versions?
In-Reply-To: <CAAtWKeGStyXKm4+o9FKJFAGNski=06RBjGSR54No+SQHYk_SFQ@mail.gmail.com>
Message-ID: <CC7F432C.4D3F4%krice@freeswitch.org>

Checkout limit_execute


On 9/19/12 9:31 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com> wrote:

> Hello,
> ?
> ? I am using limit_usage in the following format:
> ?
> <action application="bridge" data="${cond(${limit_usage(db, max_calls
> ${destination_number})} > ${MaxLimit} ?? error/user_busy :
> ${destination_mumber})}" />
> ?
> It worked at older versions (my local reference is 1.0.6), but it doesn't work
> in newer ones. It always returns zero.
> ?
> All I want is to know whether the extension already has an active call to
> decide what to do in case it is busy. What is the correct way of calling it?
> ?
> ??????????????????? 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Sep 19 18:40:06 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Sep 2012 09:40:06 -0500
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <B9B36BB1-1067-4043-8A39-625EE4359261@kavun.ch>
Message-ID: <CC7F4376.4D3F5%krice@freeswitch.org>

You can call an extension whatever you want... It doesn't have to be a
number... But using the separate context allows you to keep the destination
number field in the CDR something same like that actual destination
number...


On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:

> Is there a way to address an extension without assigning it to a
> destination_number? I don't want the extension to be reachable directly and
> would gladly avoid creating a dedicated context.
> Can I transfer a call to an extension that doesn't have a destination_number?
> 
> Best,
> Emrah
> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
> 
>> Easy indeed, but I wanted to avoid it thinking there was some other
>> alternative. 
>> Thanks for your quick response, will use it.
>> 
>> Cheers,
>> Emrah
>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
>> 
>>> This is easy to handle... Do it as 2 different extensions and after you
>>> collect the PIN, use the transfer application to transfer the call to a pin
>>> validating extension... Problem fixed with a very simple solution
>>> 
>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>>> 
>>>> Please bypass the inline="true"? A fantasy of mine.
>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>>>> 
>>>>> Hi all,
>>>>> 
>>>>> My example below cannot work because FS does not store the value of ${pin}
>>>>> until the call goes into execution state. While keeping it simple, what
>>>>> would
>>>>> be your take on this? Is there a way to instruct FS to start executing at
>>>>> some point and discovering the dialplan along the way?
>>>>> 
>>>>> <extension name="pin test">
>>>>> <condition field="destination_number" expression="^111$" break="never">
>>>>> <action application="answer"/>
>>>>> <action application="sleep" data="1000"/>
>>>>> <action application="read" data="4 4
>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
>>>>> #,*"
>>>>> inline="true" />
>>>>> </condition>
>>>>> <condition field="${pin}" expression="^1234$">
>>>>> <action application="transfer" data ="3000 XML local" />
>>>>> </condition>
>>>>> <condition field="${pin}" expression="^\d{4}$">
>>>>> <action application="hangup" />
>>>>> </condition>
>>>>> </extension>
>>>>> 
>>>>> Thanks and all the best, as always.
>>>>> 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
>>> 
>>> -- 
>>> 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
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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




From tnsampaio at bsd.com.br  Wed Sep 19 18:48:13 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Wed, 19 Sep 2012 11:48:13 -0300
Subject: [Freeswitch-users] ERROR: sofia.c:2491 Error Creating SIP UA for
	profile: internal
Message-ID: <5059DB2D.8020406@bsd.com.br>

Hi All!

Im writing an application to manage FS via mod_xml_curl.
Im generating sofia.conf just generating default configuration joining 
../sip_profiles/* on my xml.
Its an example:

<?xml version="1.0" encoding="UTF8"?>
<document type="freeswitch/xml">
   <section name="configuration">
     <configuration name="sofia.conf" description="Sessao de 
configuracao sofia.conf">
       <global_settings>
         <log-level value="0"/>
         <debug-presence value="0"/>
       </global_settings>
       <profiles>
         <profile name="external">
           <gateways>
           </gateways>
           <aliases/>
           <domains>
             <domain name="all" alias="false" parse="true"/>
           </domains>
           <settings>
             <param debug="0"/>
             <param sip-trace="no"/>
             <param sip-capture="no"/>
             <param rfc2833-pt="101"/>
             <param sip-port="5080"/>
             <param dialplan="XML"/>
             <param context="public"/>
             <param dtmf-duration="2000"/>
             <param inbound-codec-prefs="$${global_codec_prefs}"/>
             <param outbound-codec-prefs="$${outbound_codec_prefs}"/>
             <param hold-music="$${hold_music}"/>
             <param rtp-timer-name="soft"/>
             <param local-network-acl="localnet.auto"/>
             <param manage-presence="auto"/>
             <param inbound-codec-negotiation="generous"/>
             <param nonce-ttl="60"/>
             <param auth-calls="false"/>
             <param rtp-ip="$${local_ip_v4}"/>
             <param sip-ip="$${local_ip_v4}"/>
             <param ext-rtp-ip="auto-nat"/>
             <param ext-sip-ip="auto-nat"/>
             <param rtp-timeout-sec="300"/>
             <param rtp-hold-timeout-sec="1800"/>
             <param tls="$${external_ssl_enable}"/>
             <param tls-bind-params="transport=tls"/>
             <param tls-sip-port="$${external_tls_port}"/>
             <param tls-cert-dir="$${external_ssl_dir}"/>
             <param tls-version="$${sip_tls_version}"/>
           </settings>
         </profile>
         <profile name="internal">
           <aliases/>
           <domains>
             <domain name="all" alias="true" parse="false"/>
           </domains>
           <settings>
             <param debug="0"/>
             <param sip-trace="no"/>
             <param sip-capture="no"/>
             <param watchdog-enabled="no"/>
             <param watchdog-step-timeout="30000"/>
             <param watchdog-event-timeout="30000"/>
             <param log-auth-failures="true"/>
             <param forward-unsolicited-mwi-notify="true"/>
             <param context="padrao"/>
             <param sip-port="$${internal_sip_port}"/>
             <param dialplan="XML"/>
             <param dtmf-duration="2000"/>
             <param inbound-codec-prefs="$${global_codec_prefs}"/>
             <param outbound-codec-prefs="$${outbound_codec_prefs}"/>
             <param hold-music="$${hold_music}"/>
             <param rtp-timer-name="soft"/>
             <param rtp-ip="$${local_ip_v4}"/>
             <param sip-ip="$${local_ip_v4}"/>
             <param apply-nat-acl="nat.auto"/>
             <param apply-inbound-acl="domain"/>
             <param local-network-acl="localnet.auto"/>
             <param presence-hosts="$${domain},$${local_ip_v4}"/>
             <param presence-privacy="$${presence_privacy}"/>
             <param inbound-codec-negotiation="generous"/>
             <param tls="$${internal_ssl_enable}"/>
             <param tls-bind-params="transport=tls"/>
             <param tls-sip-port="$${internal_tls_port}"/>
             <param tls-cert-dir="$${external_ssl_dir}"/>
             <param tls-version="$${sip_tls_version}"/>
             <param tls-verify-date="true"/>
             <param tls-verify-policy="none"/>
             <param tls-verify-depth="2"/>
             <param tls-verify-in-subjects=""/>
             <param inbound-late-negotiation="true"/>
             <param inbound-zrtp-passthru="true"/>
             <param nonce-ttl="60"/>
             <param auth-calls="$${internal_auth_calls}"/>
             <param inbound-reg-force-matching-username="true"/>
             <param auth-all-packets="false"/>
             <param ext-rtp-ip="auto-nat"/>
             <param ext-sip-ip="auto-nat"/>
             <param rtp-timeout-sec="300"/>
             <param rtp-hold-timeout-sec="1800"/>
             <param force-register-domain="$${domain}"/>
             <param force-subscription-domain="$${domain}"/>
             <param force-register-db-domain="$${domain}"/>
             <param challenge-realm="auto_from"/>
           </settings>
         </profile>
       </profiles>
     </configuration>
   </section>
</document>

but when i do reload mod_sofia i got this:

2012-09-19 11:42:52.548225 [CONSOLE] sofia.c:1620 MSG Thread Ended
2012-09-19 11:42:52.548225 [CONSOLE] switch_loadable_module.c:1814 
mod_sofia unloaded.
2012-09-19 11:42:53.268217 [CONSOLE] mod_xml_curl.c:318 XML response is 
in /tmp/41a9f34a-6802-e211-9d5d-ced1711242b1.tmp.xml
2012-09-19 11:42:53.348238 [CONSOLE] mod_xml_curl.c:318 XML response is 
in /tmp/c8a3004b-6802-e211-9d5d-ced1711242b1.tmp.xml
2012-09-19 11:42:53.348238 [INFO] mod_enum.c:871 ENUM Reloaded
2012-09-19 11:42:53.428240 [INFO] mod_sofia.c:5652 Starting initial 
message thread.
2012-09-19 11:42:53.428240 [CONSOLE] sofia.c:1602 MSG Thread 0 Started
2012-09-19 11:42:53.488212 [CONSOLE] mod_xml_curl.c:318 XML response is 
in /tmp/7fbd0d4b-6802-e211-9d5d-ced1711242b1.tmp.xml
2012-09-19 11:42:53.488212 [INFO] switch_time.c:1163 Timezone reloaded 
530 definitions
2012-09-19 11:42:53.568343 [CONSOLE] mod_xml_curl.c:318 XML response is 
in /tmp/b333194b-6802-e211-9d5d-ced1711242b1.tmp.xml
2012-09-19 11:42:53.588215 [INFO] sofia.c:5281 Setting nonce TTL to 60 
seconds
2012-09-19 11:42:53.588215 [NOTICE] sofia.c:5358 Started Profile 
external [sofia_reg_external]
2012-09-19 11:42:53.668232 [CONSOLE] mod_xml_curl.c:318 XML response is 
in /tmp/42f5314b-6802-e211-9d5d-ced1711242b1.tmp.xml
2012-09-19 11:42:53.668232 [NOTICE] sofia_reg.c:2954 Added gateway 
'OTIMA' to profile 'external'
2012-09-19 11:42:53.768237 [CONSOLE] mod_xml_curl.c:318 XML response is 
in /tmp/8d68404b-6802-e211-9d5d-ced1711242b1.tmp.xml
2012-09-19 11:42:53.768237 [INFO] sofia.c:5281 Setting nonce TTL to 60 
seconds
2012-09-19 11:42:53.768237 [NOTICE] sofia.c:5358 Started Profile 
internal [sofia_reg_internal]

2012-09-19 11:42:53.768237 [ERR] sofia.c:2491 Error Creating SIP UA for 
profile: internal

2012-09-19 11:42:54.768518 [NOTICE] sofia_reg.c:415 Registering OTIMA
2012-09-19 11:42:55.288223 [CONSOLE] switch_loadable_module.c:1328 
Successfully Loaded [mod_sofia]
2012-09-19 11:42:55.288223 [NOTICE] switch_loadable_module.c:146 Adding 
Endpoint 'sofia'
2012-09-19 11:42:55.288223 [NOTICE] switch_loadable_module.c:254 Adding 
Application 'sofia_sla'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding 
API Function 'sofia'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding 
API Function 'sofia_gateway_data'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding 
API Function 'sofia_username_of'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding 
API Function 'sofia_contact'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding 
API Function 'sofia_count_reg'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding 
API Function 'sofia_dig'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:403 Adding 
Chat interface 'sip'
2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:447 Adding 
Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1001]

I searched a lot to get help, but cant find...




From lists at kavun.ch  Wed Sep 19 19:46:06 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 11:46:06 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <CC7F4376.4D3F5%krice@freeswitch.org>
References: <CC7F4376.4D3F5%krice@freeswitch.org>
Message-ID: <BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>

Naming an extension won't help for my transfer purpose.
What I think  I will do is use the same 1 extension idea, but verify the existance of ${pin} before executing the read instruction.
This way if ${pin} exists, it will execute one set of instructions, if it doesn't, it will execute the read app. Following the read app would just be an action to transfer the user to the same context and destination_number.

I hope what I tried to explain makes sense.
On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:

> You can call an extension whatever you want... It doesn't have to be a
> number... But using the separate context allows you to keep the destination
> number field in the CDR something same like that actual destination
> number...
> 
> 
> On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
> 
>> Is there a way to address an extension without assigning it to a
>> destination_number? I don't want the extension to be reachable directly and
>> would gladly avoid creating a dedicated context.
>> Can I transfer a call to an extension that doesn't have a destination_number?
>> 
>> Best,
>> Emrah
>> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
>> 
>>> Easy indeed, but I wanted to avoid it thinking there was some other
>>> alternative. 
>>> Thanks for your quick response, will use it.
>>> 
>>> Cheers,
>>> Emrah
>>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> 
>>>> This is easy to handle... Do it as 2 different extensions and after you
>>>> collect the PIN, use the transfer application to transfer the call to a pin
>>>> validating extension... Problem fixed with a very simple solution
>>>> 
>>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>>>> 
>>>>> Please bypass the inline="true"? A fantasy of mine.
>>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>>>>> 
>>>>>> Hi all,
>>>>>> 
>>>>>> My example below cannot work because FS does not store the value of ${pin}
>>>>>> until the call goes into execution state. While keeping it simple, what
>>>>>> would
>>>>>> be your take on this? Is there a way to instruct FS to start executing at
>>>>>> some point and discovering the dialplan along the way?
>>>>>> 
>>>>>> <extension name="pin test">
>>>>>> <condition field="destination_number" expression="^111$" break="never">
>>>>>> <action application="answer"/>
>>>>>> <action application="sleep" data="1000"/>
>>>>>> <action application="read" data="4 4
>>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
>>>>>> #,*"
>>>>>> inline="true" />
>>>>>> </condition>
>>>>>> <condition field="${pin}" expression="^1234$">
>>>>>> <action application="transfer" data ="3000 XML local" />
>>>>>> </condition>
>>>>>> <condition field="${pin}" expression="^\d{4}$">
>>>>>> <action application="hangup" />
>>>>>> </condition>
>>>>>> </extension>
>>>>>> 
>>>>>> Thanks and all the best, as always.
>>>>>> 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
>>>> 
>>>> -- 
>>>> 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
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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



From anthony.minessale at gmail.com  Wed Sep 19 19:51:58 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 19 Sep 2012 10:51:58 -0500
Subject: [Freeswitch-users] How to disable adding of rows with LOSE_RACE
 case ?
In-Reply-To: <SNT127-W651BB345720F0D2ADD03D6D5940@phx.gbl>
References: <SNT127-W56C7748D1F39A7ADF71646D5940@phx.gbl>
	<CALNZuwkaWW9tAz3G8427rfUEcJNFU1_1QUYHG3LvD86gAGDqZQ@mail.gmail.com>
	<SNT127-W651BB345720F0D2ADD03D6D5940@phx.gbl>
Message-ID: <CAKbxfG9HPGP6KvN+6HmhXuobuiJ8o75JZb6G+SFzhSFa7rc3mg@mail.gmail.com>

If you are adventurous you could update to latest GIT HEAD and try
exporting, or globally setting, or putting in {} on the originate the
variable skip_cdr_causes=LOSE_RACE

<action application="bridge"
data="{skip_cdr_causes=LOSE_RACE}user/1234,sofia/internal/foo at bar.com"/>



On Tue, Sep 18, 2012 at 4:51 PM, Yuriy Nasida <nasida at live.ru> wrote:

>  Avi,
>
> Yes you are correct. Cdrs via xml_cdr. I have done this from httpd side
> already. I tried to create some method with process_cdr false but without
> any luck.
>
> Thanks.
>
> ------------------------------
> From: avi at avimarcus.net
> Date: Tue, 18 Sep 2012 20:14:01 +0300
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] How to disable adding of rows with
> LOSE_RACE case ?
>
>
> If you use xml_cdr or json_cdr that posts to a database, you can just drop
> those CDRs.
>
> I wonder if you can use a hangup hook of a sort (before reporting!) to
> set process_cdr to false, or a_only...
>
>
> -Avi
>
>
> On Tue, Sep 18, 2012 at 5:11 PM, Yuriy Nasida <nasida at live.ru> wrote:
>
>  Hello guys!
>
> I use many endpoints in my bridge appl. So in cdr I see one good row (for
> ext which picks up call) and  many rows with "LOSE_RACE" for others ext
> from bridge string. How can I disable adding of rows with LOSE_RACE ?
>
> Please advice.
> 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 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Wed Sep 19 20:09:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Sep 2012 09:09:31 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today: Repro SIP proxy
Message-ID: <CAKzWOxW6DuSMWjzp26cHaWRXCvNc0FhM1uoh_qQrui0hQ5sT=A@mail.gmail.com>

Hello all,

Please join us for the conference call at 1PM EDT, 10AM PDT:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19

Scott and Daniel will be discussing the Repro SIP proxy and the ReSIProcate
SIP stack. 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 msc at freeswitch.org  Wed Sep 19 20:22:56 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Sep 2012 09:22:56 -0700
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>
References: <CC7F4376.4D3F5%krice@freeswitch.org>
	<BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>
Message-ID: <CAKzWOxU0ZsvW_f4Dp_iGZZGVe8_1ke-YZYTpK55Jcc5rbEvOHQ@mail.gmail.com>

There is no valid case not to use transfer (or even execute_extension) in
this scenario. Create a new file: conf/dialplan/check_pin.xml:

<include>
  <context name="utility">
    <extension name="Check PIN">
      <condition field="destination_number" expression="^check_pin$"/>
      <condition field="${pin}" expression="^1234$">
        <action application="transfer" data ="3000 XML local" />
        <anti-action application="hangup" data="Login Failed"/>
      </condition>
    </extension>
  </context>
</include>

Then just do this in your original extension that has the read app (note I
changed the name of the extension for clarity):

<extension name="Get PIN from caller">
  <condition field="destination_number" expression="^111$" break="never">
    <action application="answer"/>
    <action application="sleep" data="1000"/>
    <action application="read" data="4 4
$${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
#,*"/>
    <action application="log" data="INFO user entered '${pin}'"/>
    <action application="transfer" data="check_pin XML utility"/>
  </condition>
</extension>

That's all there is to it. If this method seems weird, or unusual, or
abstract then too bad! :) It's a great way to keep things relatively simple
while ensuring a level of security.

-MC

On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:

> Naming an extension won't help for my transfer purpose.
> What I think  I will do is use the same 1 extension idea, but verify the
> existance of ${pin} before executing the read instruction.
> This way if ${pin} exists, it will execute one set of instructions, if it
> doesn't, it will execute the read app. Following the read app would just be
> an action to transfer the user to the same context and destination_number.
>
> I hope what I tried to explain makes sense.
> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> > You can call an extension whatever you want... It doesn't have to be a
> > number... But using the separate context allows you to keep the
> destination
> > number field in the CDR something same like that actual destination
> > number...
> >
> >
> > On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
> >
> >> Is there a way to address an extension without assigning it to a
> >> destination_number? I don't want the extension to be reachable directly
> and
> >> would gladly avoid creating a dedicated context.
> >> Can I transfer a call to an extension that doesn't have a
> destination_number?
> >>
> >> Best,
> >> Emrah
> >> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
> >>
> >>> Easy indeed, but I wanted to avoid it thinking there was some other
> >>> alternative.
> >>> Thanks for your quick response, will use it.
> >>>
> >>> Cheers,
> >>> Emrah
> >>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
> >>>
> >>>> This is easy to handle... Do it as 2 different extensions and after
> you
> >>>> collect the PIN, use the transfer application to transfer the call to
> a pin
> >>>> validating extension... Problem fixed with a very simple solution
> >>>>
> >>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> >>>>
> >>>>> Please bypass the inline="true"? A fantasy of mine.
> >>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> >>>>>
> >>>>>> Hi all,
> >>>>>>
> >>>>>> My example below cannot work because FS does not store the value of
> ${pin}
> >>>>>> until the call goes into execution state. While keeping it simple,
> what
> >>>>>> would
> >>>>>> be your take on this? Is there a way to instruct FS to start
> executing at
> >>>>>> some point and discovering the dialplan along the way?
> >>>>>>
> >>>>>> <extension name="pin test">
> >>>>>> <condition field="destination_number" expression="^111$"
> break="never">
> >>>>>> <action application="answer"/>
> >>>>>> <action application="sleep" data="1000"/>
> >>>>>> <action application="read" data="4 4
> >>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin
> 10000
> >>>>>> #,*"
> >>>>>> inline="true" />
> >>>>>> </condition>
> >>>>>> <condition field="${pin}" expression="^1234$">
> >>>>>> <action application="transfer" data ="3000 XML local" />
> >>>>>> </condition>
> >>>>>> <condition field="${pin}" expression="^\d{4}$">
> >>>>>> <action application="hangup" />
> >>>>>> </condition>
> >>>>>> </extension>
> >>>>>>
> >>>>>> Thanks and all the best, as always.
> >>>>>> 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
> >>>>
> >>>> --
> >>>> 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
> >>>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 19 20:36:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Sep 2012 09:36:27 -0700
Subject: [Freeswitch-users] ESL API command and ringback
In-Reply-To: <CACSMF4wT1qpB1R-exHx1QPmkHvOy+CP1+kTLQ=Aa3JKeRke+Bw@mail.gmail.com>
References: <CACSMF4wT1qpB1R-exHx1QPmkHvOy+CP1+kTLQ=Aa3JKeRke+Bw@mail.gmail.com>
Message-ID: <CAKzWOxU-kadGhzQZDX1yXzZKyhN_bkB4bGeTFpCALmnF5fbcCg@mail.gmail.com>

I think you're correct to use ringback and ring_ready together. Set the
ringback variable and then execute the ring_ready app and you should be
golden.
-MC

On Tue, Sep 18, 2012 at 11:01 PM, Peter van Raamsdonk <peetzer at gmail.com>wrote:

> Hi again [?],
>
> Simple question,
>
> In the dialplan there is a Local_Extension to handle calls.
>
> In this example, you have an action;
>
> *<action application="set" data="ringback=${us-ring}"/>*
>
> As api command (from ESL java outbound example) I put this like;
>
>
> String result = doAction(
> "global_getvar", "us-ring");// using CLI (inbound), result  *
> %(2000,4000,440,480)* as defined in vars.xml
>
> SendMsg sendMsg =
> *new* SendMsg();
>
> sendMsg.addCallCommand(
> "execute");
>
> sendMsg.addExecuteAppName(
> "set");
>
> sendMsg.addExecuteAppArg(
> "ringback=" + result);
>
> but I don't hear sound. I don't know if the string
> argument "${us-string}", can be parsed directly.
>
> I tried everything but no sound.
>
> Only thing that I got working is by using "execute" and "ring_ready"
>
> What I'm doing wrong with ringback argument?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 19 20:46:29 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Sep 2012 09:46:29 -0700
Subject: [Freeswitch-users] ERROR: sofia.c:2491 Error Creating SIP UA
 for profile: internal
In-Reply-To: <5059DB2D.8020406@bsd.com.br>
References: <5059DB2D.8020406@bsd.com.br>
Message-ID: <CAKzWOxWmAnjVaOXuUYsG2_Y+bue1ZektMWkzmgHcUHYjNokY3g@mail.gmail.com>

> 2012-09-19 11:42:53.768237 [ERR] sofia.c:2491 Error Creating SIP UA for
> profile: internal
>
>
This is frequently a case of something else already using port 5060. Check
out this page for tips on finding out what is using that port:
http://wiki.freeswitch.org/wiki/Sofia-SIP#What_if_these_commands_don.27t_work_for_me.3F

-MC
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From tnsampaio at bsd.com.br  Wed Sep 19 21:36:16 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Wed, 19 Sep 2012 14:36:16 -0300
Subject: [Freeswitch-users] ERROR: sofia.c:2491 Error Creating SIP UA
 for profile: internal
In-Reply-To: <CAKzWOxWmAnjVaOXuUYsG2_Y+bue1ZektMWkzmgHcUHYjNokY3g@mail.gmail.com>
References: <5059DB2D.8020406@bsd.com.br>
	<CAKzWOxWmAnjVaOXuUYsG2_Y+bue1ZektMWkzmgHcUHYjNokY3g@mail.gmail.com>
Message-ID: <505A0290.2050306@bsd.com.br>

Yes, you're correct.
Now, when i disable profile internal, external profile is listening only 
on port 5060..
and if you look my xml you will see <param sip-port="5060"> its 
completly wrong...
So when it loads external profile and didnt match any valid parameter, 
it start listening on port 5060...
when i change xml params to : <param name="sip-port" value="5060"> it 
started to work...

But FS must show a better error msg like: "cant listen to port XXXX" so 
message will be much more easy to understand!

Thx for your help....



Em 19/09/2012 13:46, Michael Collins escreveu:
>
>     2012-09-19 11:42:53.768237 [ERR] sofia.c:2491 Error Creating SIP
>     UA for
>     profile: internal
>
>
> This is frequently a case of something else already using port 5060. 
> Check out this page for tips on finding out what is using that port:
> http://wiki.freeswitch.org/wiki/Sofia-SIP#What_if_these_commands_don.27t_work_for_me.3F
>
> -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  Wed Sep 19 21:40:56 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 19 Sep 2012 12:40:56 -0500
Subject: [Freeswitch-users] ERROR: sofia.c:2491 Error Creating SIP UA
 for profile: internal
In-Reply-To: <505A0290.2050306@bsd.com.br>
References: <5059DB2D.8020406@bsd.com.br>
	<CAKzWOxWmAnjVaOXuUYsG2_Y+bue1ZektMWkzmgHcUHYjNokY3g@mail.gmail.com>
	<505A0290.2050306@bsd.com.br>
Message-ID: <CAKbxfG8qOAcnDodVkH=mTWec=JMa_zGgc9CkRwaLo5gBHtv0xg@mail.gmail.com>

Commit was already pushed this morning to spell it out better.


On Wed, Sep 19, 2012 at 12:36 PM, Tiago N. Sampaio <tnsampaio at bsd.com.br>wrote:

>  Yes, you're correct.
> Now, when i disable profile internal, external profile is listening only
> on port 5060..
> and if you look my xml you will see <param sip-port="5060"> its completly
> wrong...
> So when it loads external profile and didnt match any valid parameter, it
> start listening on port 5060...
> when i change xml params to : <param name="sip-port" value="5060"> it
> started to work...
>
> But FS must show a better error msg like: "cant listen to port XXXX" so
> message will be much more easy to understand!
>
> Thx for your help....
>
>
>
> Em 19/09/2012 13:46, Michael Collins escreveu:
>
>
>  2012-09-19 11:42:53.768237 [ERR] sofia.c:2491 Error Creating SIP UA for
>> profile: internal
>>
>>
>  This is frequently a case of something else already using port 5060.
> Check out this page for tips on finding out what is using that port:
>
> http://wiki.freeswitch.org/wiki/Sofia-SIP#What_if_these_commands_don.27t_work_for_me.3F
>
> -MC
>
>
> _________________________________________________________________________
> 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
>
>


-- 
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 gavin.henry at gmail.com  Wed Sep 19 21:54:25 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Wed, 19 Sep 2012 18:54:25 +0100
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CAPerZE_XRytyQ25xr0cQ42Dw0M+KVSqF+BT+6G57SGwFEEOCfA@mail.gmail.com>
References: <CAPerZE_XRytyQ25xr0cQ42Dw0M+KVSqF+BT+6G57SGwFEEOCfA@mail.gmail.com>
Message-ID: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>

Hi Campbell,

I run SureVoIP and would love to do this. We've tired to get a quote before
but didn't get anywhere. Happy for all to go to the project.

Thanks.

On Wednesday, 12 September 2012, Campbell Steven wrote:

> Hi en_UK FreeSWITCH users,
>
> I'm trying to get a full prompt set (every prompt currently recorded
> by Callie including the ztrp prompts etc..) organised. The talent
> would be female and prompts would be recorded through GM Voices and
> therefore easy to get matching custom recordings for your applications
> into the future. The prompt set would then be given to the FreeSWITCH
> project for distribution as per the existing prompts.
>
> Anyone who is willing to help me out with funding this or has any
> questions please contact me off-list as I'd really like to get this
> sorted. Every little bit counts here so even if you just want to drop
> a few $ in it all adds up.
>
> Thanks
>
> Campbell
>
> _________________________________________________________________________
> 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
>


-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk
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From krice at freeswitch.org  Wed Sep 19 21:59:11 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Sep 2012 12:59:11 -0500
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
Message-ID: <CC7F721F.4D40B%krice@freeswitch.org>

Hey Guys,

One thing we could try, is to get OSTAG involved here... Get a quote from GM
Voices and then get several of you en_UK guys to donated a few quid to OSTAG
to cover the cost and OSTAG orders it and donates it opensource to
FreeSWITCH everyone wins

K


On 9/19/12 12:54 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:

> Hi Campbell,
> 
> I run SureVoIP and would love to do this. We've tired to get a quote before
> but didn't get anywhere. Happy for all to go to the project.?
> 
> Thanks.?
> 
> On Wednesday, 12 September 2012, Campbell Steven  wrote:
>> Hi en_UK FreeSWITCH users,
>> 
>> I'm trying to get a full prompt set (every prompt currently recorded
>> by Callie including the ztrp prompts etc..) organised. The talent
>> would be female and prompts would be recorded through GM Voices and
>> therefore easy to get matching custom recordings for your applications
>> into the future. The prompt set would then be given to the FreeSWITCH
>> project for distribution as per the existing prompts.
>> 
>> Anyone who is willing to help me out with funding this or has any
>> questions please contact me off-list as I'd really like to get this
>> sorted. Every little bit counts here so even if you just want to drop
>> a few $ in it all adds up.
>> 
>> Thanks
>> 
>> Campbell
>> 
>> _________________________________________________________________________
>> 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
> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From avi at avimarcus.net  Wed Sep 19 22:05:38 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Sep 2012 21:05:38 +0300
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CC7F721F.4D40B%krice@freeswitch.org>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
Message-ID: <CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>

Could we get a quote and a pledge list, to see how short we are?

-Avi


On Wed, Sep 19, 2012 at 8:59 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Hey Guys,
>
> One thing we could try, is to get OSTAG involved here... Get a quote from
> GM Voices and then get several of you en_UK guys to donated a few quid to
> OSTAG to cover the cost and OSTAG orders it and donates it opensource to
> FreeSWITCH everyone wins
>
> K
>
>
>
> On 9/19/12 12:54 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:
>
> Hi Campbell,
>
> I run SureVoIP and would love to do this. We've tired to get a quote
> before but didn't get anywhere. Happy for all to go to the project.
>
> Thanks.
>
> On Wednesday, 12 September 2012, Campbell Steven  wrote:
>
> Hi en_UK FreeSWITCH users,
>
> I'm trying to get a full prompt set (every prompt currently recorded
> by Callie including the ztrp prompts etc..) organised. The talent
> would be female and prompts would be recorded through GM Voices and
> therefore easy to get matching custom recordings for your applications
> into the future. The prompt set would then be given to the FreeSWITCH
> project for distribution as per the existing prompts.
>
> Anyone who is willing to help me out with funding this or has any
> questions please contact me off-list as I'd really like to get this
> sorted. Every little bit counts here so even if you just want to drop
> a few $ in it all adds up.
>
> Thanks
>
> Campbell
>
> _________________________________________________________________________
> 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
>
>
>
> --
> 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 lists at kavun.ch  Wed Sep 19 22:20:19 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 14:20:19 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <CAKzWOxU0ZsvW_f4Dp_iGZZGVe8_1ke-YZYTpK55Jcc5rbEvOHQ@mail.gmail.com>
References: <CC7F4376.4D3F5%krice@freeswitch.org>
	<BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>
	<CAKzWOxU0ZsvW_f4Dp_iGZZGVe8_1ke-YZYTpK55Jcc5rbEvOHQ@mail.gmail.com>
Message-ID: <7D02114C-339E-4F2F-B0B1-E5ED979E251E@kavun.ch>

There is a valid case to me, and here it is Sir. :)
<extension name="pin test">
<condition field="${pin}" expression="^1234$" break="on-true">
<action application="log" data="INFO Pin is ${pin}" />
<action application="transfer" data="3000 XML local" />
</condition>
<condition field="${pin}" expression="^\d{4}" break="never">
<action application="playback" data="conference/conf-bad-pin.wav" />
<action application="unset" data="pin" />
</condition>
<condition field="destination_number" expression="^111$" break="never">
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="read" data="4 4 conference/conf-pin.wav pin 10000 #,*" />
<action application="transfer" data="111 XML local" />
</condition>
</extension>

I think I'm starting to get the hang of this thing!

E

On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org> wrote:

> There is no valid case not to use transfer (or even execute_extension) in this scenario. Create a new file: conf/dialplan/check_pin.xml:
> 
> <include>
>   <context name="utility">
>     <extension name="Check PIN">
>       <condition field="destination_number" expression="^check_pin$"/>
>       <condition field="${pin}" expression="^1234$">
>         <action application="transfer" data ="3000 XML local" />
>         <anti-action application="hangup" data="Login Failed"/>
>       </condition>
>     </extension>
>   </context>
> </include>
> 
> Then just do this in your original extension that has the read app (note I changed the name of the extension for clarity):
> 
> <extension name="Get PIN from caller">
>   <condition field="destination_number" expression="^111$" break="never">
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"/>
>     <action application="log" data="INFO user entered '${pin}'"/>
>     <action application="transfer" data="check_pin XML utility"/>     
>   </condition>
> </extension>
> 
> That's all there is to it. If this method seems weird, or unusual, or abstract then too bad! :) It's a great way to keep things relatively simple while ensuring a level of security.
> 
> -MC
> 
> On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
> Naming an extension won't help for my transfer purpose.
> What I think  I will do is use the same 1 extension idea, but verify the existance of ${pin} before executing the read instruction.
> This way if ${pin} exists, it will execute one set of instructions, if it doesn't, it will execute the read app. Following the read app would just be an action to transfer the user to the same context and destination_number.
> 
> I hope what I tried to explain makes sense.
> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
> 
> > You can call an extension whatever you want... It doesn't have to be a
> > number... But using the separate context allows you to keep the destination
> > number field in the CDR something same like that actual destination
> > number...
> >
> >
> > On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
> >
> >> Is there a way to address an extension without assigning it to a
> >> destination_number? I don't want the extension to be reachable directly and
> >> would gladly avoid creating a dedicated context.
> >> Can I transfer a call to an extension that doesn't have a destination_number?
> >>
> >> Best,
> >> Emrah
> >> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
> >>
> >>> Easy indeed, but I wanted to avoid it thinking there was some other
> >>> alternative.
> >>> Thanks for your quick response, will use it.
> >>>
> >>> Cheers,
> >>> Emrah
> >>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
> >>>
> >>>> This is easy to handle... Do it as 2 different extensions and after you
> >>>> collect the PIN, use the transfer application to transfer the call to a pin
> >>>> validating extension... Problem fixed with a very simple solution
> >>>>
> >>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> >>>>
> >>>>> Please bypass the inline="true"? A fantasy of mine.
> >>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> >>>>>
> >>>>>> Hi all,
> >>>>>>
> >>>>>> My example below cannot work because FS does not store the value of ${pin}
> >>>>>> until the call goes into execution state. While keeping it simple, what
> >>>>>> would
> >>>>>> be your take on this? Is there a way to instruct FS to start executing at
> >>>>>> some point and discovering the dialplan along the way?
> >>>>>>
> >>>>>> <extension name="pin test">
> >>>>>> <condition field="destination_number" expression="^111$" break="never">
> >>>>>> <action application="answer"/>
> >>>>>> <action application="sleep" data="1000"/>
> >>>>>> <action application="read" data="4 4
> >>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
> >>>>>> #,*"
> >>>>>> inline="true" />
> >>>>>> </condition>
> >>>>>> <condition field="${pin}" expression="^1234$">
> >>>>>> <action application="transfer" data ="3000 XML local" />
> >>>>>> </condition>
> >>>>>> <condition field="${pin}" expression="^\d{4}$">
> >>>>>> <action application="hangup" />
> >>>>>> </condition>
> >>>>>> </extension>
> >>>>>>
> >>>>>> Thanks and all the best, as always.
> >>>>>> 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
> >>>>
> >>>> --
> >>>> 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
> >>>
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ssinyagin at yahoo.com  Wed Sep 19 22:39:22 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 19 Sep 2012 11:39:22 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today: Repro SIP
	proxy
In-Reply-To: <CAKzWOxW6DuSMWjzp26cHaWRXCvNc0FhM1uoh_qQrui0hQ5sT=A@mail.gmail.com>
References: <CAKzWOxW6DuSMWjzp26cHaWRXCvNc0FhM1uoh_qQrui0hQ5sT=A@mail.gmail.com>
Message-ID: <1348079962.78424.YahooMailNeo@web39303.mail.mud.yahoo.com>

great, it was pretty interesting.

what would be really cool is if joining/leaving beeps would be suppressed




>________________________________
> From: Michael Collins <msc at freeswitch.org>
>To: freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org 
>Sent: Wednesday, September 19, 2012 6:09 PM
>Subject: [Freeswitch-users] FreeSWITCH Conference Call Today: Repro SIP proxy
> 
>
>Hello all,
>
>Please join us for the conference call at 1PM EDT, 10AM PDT:
>
>http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19
>
>Scott and Daniel will be discussing the Repro SIP proxy and the ReSIProcate SIP stack. Talk to you soon.
>
>
-------------- next part --------------
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From gavin.henry at gmail.com  Wed Sep 19 23:54:52 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Wed, 19 Sep 2012 20:54:52 +0100
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CC7F721F.4D40B%krice@freeswitch.org>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
Message-ID: <CAA8_NKBh0SdvQSnvbjdCSgTx5iLnjWCUYWHKPfyU3fSe8w7xcw@mail.gmail.com>

> Hey Guys,
>
> One thing we could try, is to get OSTAG involved here... Get a quote from GM
> Voices and then get several of you en_UK guys to donated a few quid to OSTAG
> to cover the cost and OSTAG orders it and donates it opensource to
> FreeSWITCH everyone wins

Sounds great. We really need a "hash key".

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


From ssinyagin at yahoo.com  Wed Sep 19 23:57:04 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 19 Sep 2012 12:57:04 -0700 (PDT)
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <7D02114C-339E-4F2F-B0B1-E5ED979E251E@kavun.ch>
References: <CC7F4376.4D3F5%krice@freeswitch.org>
	<BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>
	<CAKzWOxU0ZsvW_f4Dp_iGZZGVe8_1ke-YZYTpK55Jcc5rbEvOHQ@mail.gmail.com>
	<7D02114C-339E-4F2F-B0B1-E5ED979E251E@kavun.ch>
Message-ID: <1348084624.89607.YahooMailNeo@web39301.mail.mud.yahoo.com>

Emrah, I think an IVR menu would be much easier to handle and to maintain than what you;re trying to do. 

Or write a Lua script, then you're even more flexible in all the checkups. You can also look up the PIN in some external database instead of having it statically coded in XML





>________________________________
> From: Emrah <lists at kavun.ch>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Wednesday, September 19, 2012 8:20 PM
>Subject: Re: [Freeswitch-users] How to force FS to execute?
> 
>There is a valid case to me, and here it is Sir. :)
><extension name="pin test">
><condition field="${pin}" expression="^1234$" break="on-true">
><action application="log" data="INFO Pin is ${pin}" />
><action application="transfer" data="3000 XML local" />
></condition>
><condition field="${pin}" expression="^\d{4}" break="never">
><action application="playback" data="conference/conf-bad-pin.wav" />
><action application="unset" data="pin" />
></condition>
><condition field="destination_number" expression="^111$" break="never">
><action application="answer"/>
><action application="sleep" data="1000"/>
><action application="read" data="4 4 conference/conf-pin.wav pin 10000 #,*" />
><action application="transfer" data="111 XML local" />
></condition>
></extension>
>
>I think I'm starting to get the hang of this thing!
>
>E
>
>On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>> There is no valid case not to use transfer (or even execute_extension) in this scenario. Create a new file: conf/dialplan/check_pin.xml:
>> 
>> <include>
>>?  <context name="utility">
>>? ?  <extension name="Check PIN">
>>? ? ?  <condition field="destination_number" expression="^check_pin$"/>
>>? ? ?  <condition field="${pin}" expression="^1234$">
>>? ? ? ?  <action application="transfer" data ="3000 XML local" />
>>? ? ? ?  <anti-action application="hangup" data="Login Failed"/>
>>? ? ?  </condition>
>>? ?  </extension>
>>?  </context>
>> </include>
>> 
>> Then just do this in your original extension that has the read app (note I changed the name of the extension for clarity):
>> 
>> <extension name="Get PIN from caller">
>>?  <condition field="destination_number" expression="^111$" break="never">
>>? ?  <action application="answer"/>
>>? ?  <action application="sleep" data="1000"/>
>>? ?  <action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"/>
>>? ?  <action application="log" data="INFO user entered '${pin}'"/>
>>? ?  <action application="transfer" data="check_pin XML utility"/>? ? 
>>?  </condition>
>> </extension>
>> 
>> That's all there is to it. If this method seems weird, or unusual, or abstract then too bad! :) It's a great way to keep things relatively simple while ensuring a level of security.
>> 
>> -MC
>> 
>> On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
>> Naming an extension won't help for my transfer purpose.
>> What I think? I will do is use the same 1 extension idea, but verify the existance of ${pin} before executing the read instruction.
>> This way if ${pin} exists, it will execute one set of instructions, if it doesn't, it will execute the read app. Following the read app would just be an action to transfer the user to the same context and destination_number.
>> 
>> I hope what I tried to explain makes sense.
>> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
>> 
>> > You can call an extension whatever you want... It doesn't have to be a
>> > number... But using the separate context allows you to keep the destination
>> > number field in the CDR something same like that actual destination
>> > number...
>> >
>> >
>> > On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
>> >
>> >> Is there a way to address an extension without assigning it to a
>> >> destination_number? I don't want the extension to be reachable directly and
>> >> would gladly avoid creating a dedicated context.
>> >> Can I transfer a call to an extension that doesn't have a destination_number?
>> >>
>> >> Best,
>> >> Emrah
>> >> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
>> >>
>> >>> Easy indeed, but I wanted to avoid it thinking there was some other
>> >>> alternative.
>> >>> Thanks for your quick response, will use it.
>> >>>
>> >>> Cheers,
>> >>> Emrah
>> >>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
>> >>>
>> >>>> This is easy to handle... Do it as 2 different extensions and after you
>> >>>> collect the PIN, use the transfer application to transfer the call to a pin
>> >>>> validating extension... Problem fixed with a very simple solution
>> >>>>
>> >>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>> >>>>
>> >>>>> Please bypass the inline="true"? A fantasy of mine.
>> >>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>> >>>>>
>> >>>>>> Hi all,
>> >>>>>>
>> >>>>>> My example below cannot work because FS does not store the value of ${pin}
>> >>>>>> until the call goes into execution state. While keeping it simple, what
>> >>>>>> would
>> >>>>>> be your take on this? Is there a way to instruct FS to start executing at
>> >>>>>> some point and discovering the dialplan along the way?
>> >>>>>>
>> >>>>>> <extension name="pin test">
>> >>>>>> <condition field="destination_number" expression="^111$" break="never">
>> >>>>>> <action application="answer"/>
>> >>>>>> <action application="sleep" data="1000"/>
>> >>>>>> <action application="read" data="4 4
>> >>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
>> >>>>>> #,*"
>> >>>>>> inline="true" />
>> >>>>>> </condition>
>> >>>>>> <condition field="${pin}" expression="^1234$">
>> >>>>>> <action application="transfer" data ="3000 XML local" />
>> >>>>>> </condition>
>> >>>>>> <condition field="${pin}" expression="^\d{4}$">
>> >>>>>> <action application="hangup" />
>> >>>>>> </condition>
>> >>>>>> </extension>
>> >>>>>>
>> >>>>>> Thanks and all the best, as always.
>> >>>>>> 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
>> >>>>
>> >>>> --
>> >>>> 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
>> >>>
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at 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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
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>
>
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>http://www.freeswitchsolutions.com
>
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From lists at kavun.ch  Thu Sep 20 00:19:18 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 16:19:18 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <1348084624.89607.YahooMailNeo@web39301.mail.mud.yahoo.com>
References: <CC7F4376.4D3F5%krice@freeswitch.org>
	<BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>
	<CAKzWOxU0ZsvW_f4Dp_iGZZGVe8_1ke-YZYTpK55Jcc5rbEvOHQ@mail.gmail.com>
	<7D02114C-339E-4F2F-B0B1-E5ED979E251E@kavun.ch>
	<1348084624.89607.YahooMailNeo@web39301.mail.mud.yahoo.com>
Message-ID: <541D4D23-5531-43B1-8326-E7F5F31D899D@kavun.ch>

You are correct indeed.
This was just to get an idea. If I want data from an external source, there are plenty of dialplan options, including ${api func()} to be a little suicidal. :P
We can pretty much fall back on LUA, Perl, IVR, so on and so forth for anything and everything.
For the purpose of my test case, my example seems to fit the bill and does the job.

Cheers,
Emrah
On Sep 19, 2012, at 3:57 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> Emrah, I think an IVR menu would be much easier to handle and to maintain than what you;re trying to do. 
> Or write a Lua script, then you're even more flexible in all the checkups. You can also look up the PIN in some external database instead of having it statically coded in XML
> 
> 
> From: Emrah <lists at kavun.ch>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
> Sent: Wednesday, September 19, 2012 8:20 PM
> Subject: Re: [Freeswitch-users] How to force FS to execute?
> 
> There is a valid case to me, and here it is Sir. :)
> <extension name="pin test">
> <condition field="${pin}" expression="^1234$" break="on-true">
> <action application="log" data="INFO Pin is ${pin}" />
> <action application="transfer" data="3000 XML local" />
> </condition>
> <condition field="${pin}" expression="^\d{4}" break="never">
> <action application="playback" data="conference/conf-bad-pin.wav" />
> <action application="unset" data="pin" />
> </condition>
> <condition field="destination_number" expression="^111$" break="never">
> <action application="answer"/>
> <action application="sleep" data="1000"/>
> <action application="read" data="4 4 conference/conf-pin.wav pin 10000 #,*" />
> <action application="transfer" data="111 XML local" />
> </condition>
> </extension>
> 
> I think I'm starting to get the hang of this thing!
> 
> E
> 
> On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
> > There is no valid case not to use transfer (or even execute_extension) in this scenario. Create a new file: conf/dialplan/check_pin.xml:
> > 
> > <include>
> >  <context name="utility">
> >    <extension name="Check PIN">
> >      <condition field="destination_number" expression="^check_pin$"/>
> >      <condition field="${pin}" expression="^1234$">
> >        <action application="transfer" data ="3000 XML local" />
> >        <anti-action application="hangup" data="Login Failed"/>
> >      </condition>
> >    </extension>
> >  </context>
> > </include>
> > 
> > Then just do this in your original extension that has the read app (note I changed the name of the extension for clarity):
> > 
> > <extension name="Get PIN from caller">
> >  <condition field="destination_number" expression="^111$" break="never">
> >    <action application="answer"/>
> >    <action application="sleep" data="1000"/>
> >    <action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"/>
> >    <action application="log" data="INFO user entered '${pin}'"/>
> >    <action application="transfer" data="check_pin XML utility"/>    
> >  </condition>
> > </extension>
> > 
> > That's all there is to it. If this method seems weird, or unusual, or abstract then too bad! :) It's a great way to keep things relatively simple while ensuring a level of security.
> > 
> > -MC
> > 
> > On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
> > Naming an extension won't help for my transfer purpose.
> > What I think  I will do is use the same 1 extension idea, but verify the existance of ${pin} before executing the read instruction.
> > This way if ${pin} exists, it will execute one set of instructions, if it doesn't, it will execute the read app. Following the read app would just be an action to transfer the user to the same context and destination_number.
> > 
> > I hope what I tried to explain makes sense.
> > On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
> > 
> > > You can call an extension whatever you want... It doesn't have to be a
> > > number... But using the separate context allows you to keep the destination
> > > number field in the CDR something same like that actual destination
> > > number...
> > >
> > >
> > > On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
> > >
> > >> Is there a way to address an extension without assigning it to a
> > >> destination_number? I don't want the extension to be reachable directly and
> > >> would gladly avoid creating a dedicated context.
> > >> Can I transfer a call to an extension that doesn't have a destination_number?
> > >>
> > >> Best,
> > >> Emrah
> > >> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
> > >>
> > >>> Easy indeed, but I wanted to avoid it thinking there was some other
> > >>> alternative.
> > >>> Thanks for your quick response, will use it.
> > >>>
> > >>> Cheers,
> > >>> Emrah
> > >>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
> > >>>
> > >>>> This is easy to handle... Do it as 2 different extensions and after you
> > >>>> collect the PIN, use the transfer application to transfer the call to a pin
> > >>>> validating extension... Problem fixed with a very simple solution
> > >>>>
> > >>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> > >>>>
> > >>>>> Please bypass the inline="true"? A fantasy of mine.
> > >>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> > >>>>>
> > >>>>>> Hi all,
> > >>>>>>
> > >>>>>> My example below cannot work because FS does not store the value of ${pin}
> > >>>>>> until the call goes into execution state. While keeping it simple, what
> > >>>>>> would
> > >>>>>> be your take on this? Is there a way to instruct FS to start executing at
> > >>>>>> some point and discovering the dialplan along the way?
> > >>>>>>
> > >>>>>> <extension name="pin test">
> > >>>>>> <condition field="destination_number" expression="^111$" break="never">
> > >>>>>> <action application="answer"/>
> > >>>>>> <action application="sleep" data="1000"/>
> > >>>>>> <action application="read" data="4 4
> > >>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
> > >>>>>> #,*"
> > >>>>>> inline="true" />
> > >>>>>> </condition>
> > >>>>>> <condition field="${pin}" expression="^1234$">
> > >>>>>> <action application="transfer" data ="3000 XML local" />
> > >>>>>> </condition>
> > >>>>>> <condition field="${pin}" expression="^\d{4}$">
> > >>>>>> <action application="hangup" />
> > >>>>>> </condition>
> > >>>>>> </extension>
> > >>>>>>
> > >>>>>> Thanks and all the best, as always.
> > >>>>>> 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
> > >>>>
> > >>>> --
> > >>>> 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
> > >>>
> > >>
> > >>
> > >> _________________________________________________________________________
> > >> Professional FreeSWITCH Consulting Services:
> > >> consulting at freeswitch.org
> > >> http://www.freeswitchsolutions.com
> > >>
> > >> 
> > >> 
> > >>
> > >> Official FreeSWITCH Sites
> > >> http://www.freeswitch.org
> > >> http://wiki.freeswitch.org
> > >> http://www.cluecon.com
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at 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
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Sep 20 00:31:21 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 19 Sep 2012 16:31:21 -0400
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <541D4D23-5531-43B1-8326-E7F5F31D899D@kavun.ch>
References: <CC7F4376.4D3F5%krice@freeswitch.org>
	<BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>
	<CAKzWOxU0ZsvW_f4Dp_iGZZGVe8_1ke-YZYTpK55Jcc5rbEvOHQ@mail.gmail.com>
	<7D02114C-339E-4F2F-B0B1-E5ED979E251E@kavun.ch>
	<1348084624.89607.YahooMailNeo@web39301.mail.mud.yahoo.com>
	<541D4D23-5531-43B1-8326-E7F5F31D899D@kavun.ch>
Message-ID: <72162FFC-A9F0-45E3-926D-00FDBF6004E7@kavun.ch>

My example is far from being the best. Correct me if I am wrong, but abusing of the logic I am using can heavily load the dialplan processing? 
MFC's idea to execute an extension sounds pretty sexy.

Can my example have undesirable side effects? Obviously the var name can be something more unique.

Cheers
On Sep 19, 2012, at 4:19 PM, Emrah <lists at kavun.ch> wrote:

> You are correct indeed.
> This was just to get an idea. If I want data from an external source, there are plenty of dialplan options, including ${api func()} to be a little suicidal. :P
> We can pretty much fall back on LUA, Perl, IVR, so on and so forth for anything and everything.
> For the purpose of my test case, my example seems to fit the bill and does the job.
> 
> Cheers,
> Emrah
> On Sep 19, 2012, at 3:57 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> 
>> Emrah, I think an IVR menu would be much easier to handle and to maintain than what you;re trying to do. 
>> Or write a Lua script, then you're even more flexible in all the checkups. You can also look up the PIN in some external database instead of having it statically coded in XML
>> 
>> 
>> From: Emrah <lists at kavun.ch>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>> Sent: Wednesday, September 19, 2012 8:20 PM
>> Subject: Re: [Freeswitch-users] How to force FS to execute?
>> 
>> There is a valid case to me, and here it is Sir. :)
>> <extension name="pin test">
>> <condition field="${pin}" expression="^1234$" break="on-true">
>> <action application="log" data="INFO Pin is ${pin}" />
>> <action application="transfer" data="3000 XML local" />
>> </condition>
>> <condition field="${pin}" expression="^\d{4}" break="never">
>> <action application="playback" data="conference/conf-bad-pin.wav" />
>> <action application="unset" data="pin" />
>> </condition>
>> <condition field="destination_number" expression="^111$" break="never">
>> <action application="answer"/>
>> <action application="sleep" data="1000"/>
>> <action application="read" data="4 4 conference/conf-pin.wav pin 10000 #,*" />
>> <action application="transfer" data="111 XML local" />
>> </condition>
>> </extension>
>> 
>> I think I'm starting to get the hang of this thing!
>> 
>> E
>> 
>> On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> There is no valid case not to use transfer (or even execute_extension) in this scenario. Create a new file: conf/dialplan/check_pin.xml:
>>> 
>>> <include>
>>> <context name="utility">
>>>   <extension name="Check PIN">
>>>     <condition field="destination_number" expression="^check_pin$"/>
>>>     <condition field="${pin}" expression="^1234$">
>>>       <action application="transfer" data ="3000 XML local" />
>>>       <anti-action application="hangup" data="Login Failed"/>
>>>     </condition>
>>>   </extension>
>>> </context>
>>> </include>
>>> 
>>> Then just do this in your original extension that has the read app (note I changed the name of the extension for clarity):
>>> 
>>> <extension name="Get PIN from caller">
>>> <condition field="destination_number" expression="^111$" break="never">
>>>   <action application="answer"/>
>>>   <action application="sleep" data="1000"/>
>>>   <action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"/>
>>>   <action application="log" data="INFO user entered '${pin}'"/>
>>>   <action application="transfer" data="check_pin XML utility"/>    
>>> </condition>
>>> </extension>
>>> 
>>> That's all there is to it. If this method seems weird, or unusual, or abstract then too bad! :) It's a great way to keep things relatively simple while ensuring a level of security.
>>> 
>>> -MC
>>> 
>>> On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
>>> Naming an extension won't help for my transfer purpose.
>>> What I think  I will do is use the same 1 extension idea, but verify the existance of ${pin} before executing the read instruction.
>>> This way if ${pin} exists, it will execute one set of instructions, if it doesn't, it will execute the read app. Following the read app would just be an action to transfer the user to the same context and destination_number.
>>> 
>>> I hope what I tried to explain makes sense.
>>> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> 
>>>> You can call an extension whatever you want... It doesn't have to be a
>>>> number... But using the separate context allows you to keep the destination
>>>> number field in the CDR something same like that actual destination
>>>> number...
>>>> 
>>>> 
>>>> On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
>>>> 
>>>>> Is there a way to address an extension without assigning it to a
>>>>> destination_number? I don't want the extension to be reachable directly and
>>>>> would gladly avoid creating a dedicated context.
>>>>> Can I transfer a call to an extension that doesn't have a destination_number?
>>>>> 
>>>>> Best,
>>>>> Emrah
>>>>> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
>>>>> 
>>>>>> Easy indeed, but I wanted to avoid it thinking there was some other
>>>>>> alternative.
>>>>>> Thanks for your quick response, will use it.
>>>>>> 
>>>>>> Cheers,
>>>>>> Emrah
>>>>>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>>>> 
>>>>>>> This is easy to handle... Do it as 2 different extensions and after you
>>>>>>> collect the PIN, use the transfer application to transfer the call to a pin
>>>>>>> validating extension... Problem fixed with a very simple solution
>>>>>>> 
>>>>>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>>>>>>> 
>>>>>>>> Please bypass the inline="true"? A fantasy of mine.
>>>>>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>>>>>>>> 
>>>>>>>>> Hi all,
>>>>>>>>> 
>>>>>>>>> My example below cannot work because FS does not store the value of ${pin}
>>>>>>>>> until the call goes into execution state. While keeping it simple, what
>>>>>>>>> would
>>>>>>>>> be your take on this? Is there a way to instruct FS to start executing at
>>>>>>>>> some point and discovering the dialplan along the way?
>>>>>>>>> 
>>>>>>>>> <extension name="pin test">
>>>>>>>>> <condition field="destination_number" expression="^111$" break="never">
>>>>>>>>> <action application="answer"/>
>>>>>>>>> <action application="sleep" data="1000"/>
>>>>>>>>> <action application="read" data="4 4
>>>>>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
>>>>>>>>> #,*"
>>>>>>>>> inline="true" />
>>>>>>>>> </condition>
>>>>>>>>> <condition field="${pin}" expression="^1234$">
>>>>>>>>> <action application="transfer" data ="3000 XML local" />
>>>>>>>>> </condition>
>>>>>>>>> <condition field="${pin}" expression="^\d{4}$">
>>>>>>>>> <action application="hangup" />
>>>>>>>>> </condition>
>>>>>>>>> </extension>
>>>>>>>>> 
>>>>>>>>> Thanks and all the best, as always.
>>>>>>>>> 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
>>>>>>> 
>>>>>>> --
>>>>>>> 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
>>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 



From ssinyagin at yahoo.com  Thu Sep 20 00:45:53 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 19 Sep 2012 13:45:53 -0700 (PDT)
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <72162FFC-A9F0-45E3-926D-00FDBF6004E7@kavun.ch>
References: <CC7F4376.4D3F5%krice@freeswitch.org>
	<BBDAE233-4FE3-4092-8BC5-D70B31C53E59@kavun.ch>
	<CAKzWOxU0ZsvW_f4Dp_iGZZGVe8_1ke-YZYTpK55Jcc5rbEvOHQ@mail.gmail.com>
	<7D02114C-339E-4F2F-B0B1-E5ED979E251E@kavun.ch>
	<1348084624.89607.YahooMailNeo@web39301.mail.mud.yahoo.com>
	<541D4D23-5531-43B1-8326-E7F5F31D899D@kavun.ch>
	<72162FFC-A9F0-45E3-926D-00FDBF6004E7@kavun.ch>
Message-ID: <1348087553.98119.YahooMailNeo@web39306.mail.mud.yahoo.com>

by the way, "read", "play_and_get_digits", and the IVR framework itself, they all end up calling the same function, switch_ivr_read().


Also the IVR is actually designed to handle such interaction cases, so I don't see a point in making the dialplan more complex. I don't think the CPU processing time is an issue here, but you are adding a complexity to your configuration where it can be done simpler. It just makes configuration maintenance and troubleshooting more expensive.

just my two cents :)





>________________________________
> From: Emrah <lists at kavun.ch>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Wednesday, September 19, 2012 10:31 PM
>Subject: Re: [Freeswitch-users] How to force FS to execute?
> 
>My example is far from being the best. Correct me if I am wrong, but abusing of the logic I am using can heavily load the dialplan processing? 
>MFC's idea to execute an extension sounds pretty sexy.
>
>Can my example have undesirable side effects? Obviously the var name can be something more unique.
>
>Cheers
>On Sep 19, 2012, at 4:19 PM, Emrah <lists at kavun.ch> wrote:
>
>> You are correct indeed.
>> This was just to get an idea. If I want data from an external source, there are plenty of dialplan options, including ${api func()} to be a little suicidal. :P
>> We can pretty much fall back on LUA, Perl, IVR, so on and so forth for anything and everything.
>> For the purpose of my test case, my example seems to fit the bill and does the job.
>> 
>> Cheers,
>> Emrah
>> On Sep 19, 2012, at 3:57 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>> 
>>> Emrah, I think an IVR menu would be much easier to handle and to maintain than what you;re trying to do. 
>>> Or write a Lua script, then you're even more flexible in all the checkups. You can also look up the PIN in some external database instead of having it statically coded in XML
>>> 
>>> 
>>> From: Emrah <lists at kavun.ch>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>> Sent: Wednesday, September 19, 2012 8:20 PM
>>> Subject: Re: [Freeswitch-users] How to force FS to execute?
>>> 
>>> There is a valid case to me, and here it is Sir. :)
>>> <extension name="pin test">
>>> <condition field="${pin}" expression="^1234$" break="on-true">
>>> <action application="log" data="INFO Pin is ${pin}" />
>>> <action application="transfer" data="3000 XML local" />
>>> </condition>
>>> <condition field="${pin}" expression="^\d{4}" break="never">
>>> <action application="playback" data="conference/conf-bad-pin.wav" />
>>> <action application="unset" data="pin" />
>>> </condition>
>>> <condition field="destination_number" expression="^111$" break="never">
>>> <action application="answer"/>
>>> <action application="sleep" data="1000"/>
>>> <action application="read" data="4 4 conference/conf-pin.wav pin 10000 #,*" />
>>> <action application="transfer" data="111 XML local" />
>>> </condition>
>>> </extension>
>>> 
>>> I think I'm starting to get the hang of this thing!
>>> 
>>> E
>>> 
>>> On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> 
>>>> There is no valid case not to use transfer (or even execute_extension) in this scenario. Create a new file: conf/dialplan/check_pin.xml:
>>>> 
>>>> <include>
>>>> <context name="utility">
>>>>?  <extension name="Check PIN">
>>>>? ?  <condition field="destination_number" expression="^check_pin$"/>
>>>>? ?  <condition field="${pin}" expression="^1234$">
>>>>? ? ?  <action application="transfer" data ="3000 XML local" />
>>>>? ? ?  <anti-action application="hangup" data="Login Failed"/>
>>>>? ?  </condition>
>>>>?  </extension>
>>>> </context>
>>>> </include>
>>>> 
>>>> Then just do this in your original extension that has the read app (note I changed the name of the extension for clarity):
>>>> 
>>>> <extension name="Get PIN from caller">
>>>> <condition field="destination_number" expression="^111$" break="never">
>>>>?  <action application="answer"/>
>>>>?  <action application="sleep" data="1000"/>
>>>>?  <action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*"/>
>>>>?  <action application="log" data="INFO user entered '${pin}'"/>
>>>>?  <action application="transfer" data="check_pin XML utility"/>? ? 
>>>> </condition>
>>>> </extension>
>>>> 
>>>> That's all there is to it. If this method seems weird, or unusual, or abstract then too bad! :) It's a great way to keep things relatively simple while ensuring a level of security.
>>>> 
>>>> -MC
>>>> 
>>>> On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
>>>> Naming an extension won't help for my transfer purpose.
>>>> What I think? I will do is use the same 1 extension idea, but verify the existance of ${pin} before executing the read instruction.
>>>> This way if ${pin} exists, it will execute one set of instructions, if it doesn't, it will execute the read app. Following the read app would just be an action to transfer the user to the same context and destination_number.
>>>> 
>>>> I hope what I tried to explain makes sense.
>>>> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>> 
>>>>> You can call an extension whatever you want... It doesn't have to be a
>>>>> number... But using the separate context allows you to keep the destination
>>>>> number field in the CDR something same like that actual destination
>>>>> number...
>>>>> 
>>>>> 
>>>>> On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
>>>>> 
>>>>>> Is there a way to address an extension without assigning it to a
>>>>>> destination_number? I don't want the extension to be reachable directly and
>>>>>> would gladly avoid creating a dedicated context.
>>>>>> Can I transfer a call to an extension that doesn't have a destination_number?
>>>>>> 
>>>>>> Best,
>>>>>> Emrah
>>>>>> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
>>>>>> 
>>>>>>> Easy indeed, but I wanted to avoid it thinking there was some other
>>>>>>> alternative.
>>>>>>> Thanks for your quick response, will use it.
>>>>>>> 
>>>>>>> Cheers,
>>>>>>> Emrah
>>>>>>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>>>>> 
>>>>>>>> This is easy to handle... Do it as 2 different extensions and after you
>>>>>>>> collect the PIN, use the transfer application to transfer the call to a pin
>>>>>>>> validating extension... Problem fixed with a very simple solution
>>>>>>>> 
>>>>>>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>>>>>>>> 
>>>>>>>>> Please bypass the inline="true"? A fantasy of mine.
>>>>>>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>>>>>>>>> 
>>>>>>>>>> Hi all,
>>>>>>>>>> 
>>>>>>>>>> My example below cannot work because FS does not store the value of ${pin}
>>>>>>>>>> until the call goes into execution state. While keeping it simple, what
>>>>>>>>>> would
>>>>>>>>>> be your take on this? Is there a way to instruct FS to start executing at
>>>>>>>>>> some point and discovering the dialplan along the way?
>>>>>>>>>> 
>>>>>>>>>> <extension name="pin test">
>>>>>>>>>> <condition field="destination_number" expression="^111$" break="never">
>>>>>>>>>> <action application="answer"/>
>>>>>>>>>> <action application="sleep" data="1000"/>
>>>>>>>>>> <action application="read" data="4 4
>>>>>>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
>>>>>>>>>> #,*"
>>>>>>>>>> inline="true" />
>>>>>>>>>> </condition>
>>>>>>>>>> <condition field="${pin}" expression="^1234$">
>>>>>>>>>> <action application="transfer" data ="3000 XML local" />
>>>>>>>>>> </condition>
>>>>>>>>>> <condition field="${pin}" expression="^\d{4}$">
>>>>>>>>>> <action application="hangup" />
>>>>>>>>>> </condition>
>>>>>>>>>> </extension>
>>>>>>>>>> 
>>>>>>>>>> Thanks and all the best, as always.
>>>>>>>>>> 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
>>>>>>>> 
>>>>>>>> --
>>>>>>>> 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
>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.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 Sep 20 01:12:59 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Sep 2012 16:12:59 -0500
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <72162FFC-A9F0-45E3-926D-00FDBF6004E7@kavun.ch>
Message-ID: <CC7F9F8B.4D421%krice@freeswitch.org>

If its extremely heavily loaded why not write an application to handle the
pin collection and validation

K


On 9/19/12 3:31 PM, "Emrah" <lists at kavun.ch> wrote:

> My example is far from being the best. Correct me if I am wrong, but abusing
> of the logic I am using can heavily load the dialplan processing...
> MFC's idea to execute an extension sounds pretty sexy.
> 
> Can my example have undesirable side effects? Obviously the var name can be
> something more unique.
> 
> Cheers
> On Sep 19, 2012, at 4:19 PM, Emrah <lists at kavun.ch> wrote:
> 
>> You are correct indeed.
>> This was just to get an idea. If I want data from an external source, there
>> are plenty of dialplan options, including ${api func()} to be a little
>> suicidal. :P
>> We can pretty much fall back on LUA, Perl, IVR, so on and so forth for
>> anything and everything.
>> For the purpose of my test case, my example seems to fit the bill and does
>> the job.
>> 
>> Cheers,
>> Emrah
>> On Sep 19, 2012, at 3:57 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>> 
>>> Emrah, I think an IVR menu would be much easier to handle and to maintain
>>> than what you;re trying to do.
>>> Or write a Lua script, then you're even more flexible in all the checkups.
>>> You can also look up the PIN in some external database instead of having it
>>> statically coded in XML
>>> 
>>> 
>>> From: Emrah <lists at kavun.ch>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Wednesday, September 19, 2012 8:20 PM
>>> Subject: Re: [Freeswitch-users] How to force FS to execute?
>>> 
>>> There is a valid case to me, and here it is Sir. :)
>>> <extension name="pin test">
>>> <condition field="${pin}" expression="^1234$" break="on-true">
>>> <action application="log" data="INFO Pin is ${pin}" />
>>> <action application="transfer" data="3000 XML local" />
>>> </condition>
>>> <condition field="${pin}" expression="^\d{4}" break="never">
>>> <action application="playback" data="conference/conf-bad-pin.wav" />
>>> <action application="unset" data="pin" />
>>> </condition>
>>> <condition field="destination_number" expression="^111$" break="never">
>>> <action application="answer"/>
>>> <action application="sleep" data="1000"/>
>>> <action application="read" data="4 4 conference/conf-pin.wav pin 10000 #,*"
>>> />
>>> <action application="transfer" data="111 XML local" />
>>> </condition>
>>> </extension>
>>> 
>>> I think I'm starting to get the hang of this thing!
>>> 
>>> E
>>> 
>>> On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> 
>>>> There is no valid case not to use transfer (or even execute_extension) in
>>>> this scenario. Create a new file: conf/dialplan/check_pin.xml:
>>>> 
>>>> <include>
>>>> <context name="utility">
>>>>   <extension name="Check PIN">
>>>>     <condition field="destination_number" expression="^check_pin$"/>
>>>>     <condition field="${pin}" expression="^1234$">
>>>>       <action application="transfer" data ="3000 XML local" />
>>>>       <anti-action application="hangup" data="Login Failed"/>
>>>>     </condition>
>>>>   </extension>
>>>> </context>
>>>> </include>
>>>> 
>>>> Then just do this in your original extension that has the read app (note I
>>>> changed the name of the extension for clarity):
>>>> 
>>>> <extension name="Get PIN from caller">
>>>> <condition field="destination_number" expression="^111$" break="never">
>>>>   <action application="answer"/>
>>>>   <action application="sleep" data="1000"/>
>>>>   <action application="read" data="4 4
>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000
>>>> #,*"/>
>>>>   <action application="log" data="INFO user entered '${pin}'"/>
>>>>   <action application="transfer" data="check_pin XML utility"/>
>>>> </condition>
>>>> </extension>
>>>> 
>>>> That's all there is to it. If this method seems weird, or unusual, or
>>>> abstract then too bad! :) It's a great way to keep things relatively simple
>>>> while ensuring a level of security.
>>>> 
>>>> -MC
>>>> 
>>>> On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
>>>> Naming an extension won't help for my transfer purpose.
>>>> What I think  I will do is use the same 1 extension idea, but verify the
>>>> existance of ${pin} before executing the read instruction.
>>>> This way if ${pin} exists, it will execute one set of instructions, if it
>>>> doesn't, it will execute the read app. Following the read app would just be
>>>> an action to transfer the user to the same context and destination_number.
>>>> 
>>>> I hope what I tried to explain makes sense.
>>>> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>> 
>>>>> You can call an extension whatever you want... It doesn't have to be a
>>>>> number... But using the separate context allows you to keep the
>>>>> destination
>>>>> number field in the CDR something same like that actual destination
>>>>> number...
>>>>> 
>>>>> 
>>>>> On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
>>>>> 
>>>>>> Is there a way to address an extension without assigning it to a
>>>>>> destination_number? I don't want the extension to be reachable directly
>>>>>> and
>>>>>> would gladly avoid creating a dedicated context.
>>>>>> Can I transfer a call to an extension that doesn't have a
>>>>>> destination_number?
>>>>>> 
>>>>>> Best,
>>>>>> Emrah
>>>>>> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
>>>>>> 
>>>>>>> Easy indeed, but I wanted to avoid it thinking there was some other
>>>>>>> alternative.
>>>>>>> Thanks for your quick response, will use it.
>>>>>>> 
>>>>>>> Cheers,
>>>>>>> Emrah
>>>>>>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>>>>> 
>>>>>>>> This is easy to handle... Do it as 2 different extensions and after you
>>>>>>>> collect the PIN, use the transfer application to transfer the call to a
>>>>>>>> pin
>>>>>>>> validating extension... Problem fixed with a very simple solution
>>>>>>>> 
>>>>>>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>>>>>>>> 
>>>>>>>>> Please bypass the inline="true"? A fantasy of mine.
>>>>>>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>>>>>>>>> 
>>>>>>>>>> Hi all,
>>>>>>>>>> 
>>>>>>>>>> My example below cannot work because FS does not store the value of
>>>>>>>>>> ${pin}
>>>>>>>>>> until the call goes into execution state. While keeping it simple,
>>>>>>>>>> what
>>>>>>>>>> would
>>>>>>>>>> be your take on this? Is there a way to instruct FS to start
>>>>>>>>>> executing at
>>>>>>>>>> some point and discovering the dialplan along the way?
>>>>>>>>>> 
>>>>>>>>>> <extension name="pin test">
>>>>>>>>>> <condition field="destination_number" expression="^111$"
>>>>>>>>>> break="never">
>>>>>>>>>> <action application="answer"/>
>>>>>>>>>> <action application="sleep" data="1000"/>
>>>>>>>>>> <action application="read" data="4 4
>>>>>>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin
>>>>>>>>>> 10000
>>>>>>>>>> #,*"
>>>>>>>>>> inline="true" />
>>>>>>>>>> </condition>
>>>>>>>>>> <condition field="${pin}" expression="^1234$">
>>>>>>>>>> <action application="transfer" data ="3000 XML local" />
>>>>>>>>>> </condition>
>>>>>>>>>> <condition field="${pin}" expression="^\d{4}$">
>>>>>>>>>> <action application="hangup" />
>>>>>>>>>> </condition>
>>>>>>>>>> </extension>
>>>>>>>>>> 
>>>>>>>>>> Thanks and all the best, as always.
>>>>>>>>>> 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-use
>>>>>>>>> rs
>>>>>>>>> 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-user>>>>>>>>
s
>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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




From msc at freeswitch.org  Thu Sep 20 02:47:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Sep 2012 15:47:03 -0700
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
	<CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
Message-ID: <CAKzWOxVPNwFhBN6aUe7QQSyZ2qg4YN55e_5YquPUqkii-9Q6pQ@mail.gmail.com>

I'm okay with the pledge list but I think GMV is giving Campbell a
"please-don't-make-this-public" price because it is relatively low for a
full sound set. (They love the FS project, so they do cool stuff like this.)

If you want to help out please send a pledge message directly to Campbell
Steven so that he can see what the interest is. Additionally, OSTAG would
be happy to act as an agent and hold any donated funds in escrow until it's
time to pay GMV. If you have money and are ready to donate now then simply
Paypal money to donations at ostag.org and add instructions to seller saying
"UK English sounds" and we'll know what that means. If you prefer to donate
via a different method then check out our
how-to-donate<http://ostag.org/node/7>page.

Thanks!
-MC

On Wed, Sep 19, 2012 at 11:05 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Could we get a quote and a pledge list, to see how short we are?
>
> -Avi
>
>
> On Wed, Sep 19, 2012 at 8:59 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Hey Guys,
>>
>> One thing we could try, is to get OSTAG involved here... Get a quote from
>> GM Voices and then get several of you en_UK guys to donated a few quid to
>> OSTAG to cover the cost and OSTAG orders it and donates it opensource to
>> FreeSWITCH everyone wins
>>
>> K
>>
>>
>>
>> On 9/19/12 12:54 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:
>>
>> Hi Campbell,
>>
>> I run SureVoIP and would love to do this. We've tired to get a quote
>> before but didn't get anywhere. Happy for all to go to the project.
>>
>> Thanks.
>>
>> On Wednesday, 12 September 2012, Campbell Steven  wrote:
>>
>> Hi en_UK FreeSWITCH users,
>>
>> I'm trying to get a full prompt set (every prompt currently recorded
>> by Callie including the ztrp prompts etc..) organised. The talent
>> would be female and prompts would be recorded through GM Voices and
>> therefore easy to get matching custom recordings for your applications
>> into the future. The prompt set would then be given to the FreeSWITCH
>> project for distribution as per the existing prompts.
>>
>> Anyone who is willing to help me out with funding this or has any
>> questions please contact me off-list as I'd really like to get this
>> sorted. Every little bit counts here so even if you just want to drop
>> a few $ in it all adds up.
>>
>> Thanks
>>
>> Campbell
>>
>> _________________________________________________________________________
>> 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
>>
>>
>>
>> --
>> 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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 20 02:51:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Sep 2012 15:51:27 -0700
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <CC7F9F8B.4D421%krice@freeswitch.org>
References: <72162FFC-A9F0-45E3-926D-00FDBF6004E7@kavun.ch>
	<CC7F9F8B.4D421%krice@freeswitch.org>
Message-ID: <CAKzWOxXc6+UFSSNXm25Vc-RC1+h0tD4B8iQY-HmjXriOeq0B7A@mail.gmail.com>

Don't we have a PIN validation routine in mod_voicemail? Alternatively, one
could write a semi-generic PIN validation Lua script relatively easily. If
no one does it prior to me getting to chapter 7 of the FS book re-write
then I will consider doing it as an example script.

-MC

On Wed, Sep 19, 2012 at 2:12 PM, Ken Rice <krice at freeswitch.org> wrote:

> If its extremely heavily loaded why not write an application to handle the
> pin collection and validation
>
> K
>
>
> On 9/19/12 3:31 PM, "Emrah" <lists at kavun.ch> wrote:
>
> > My example is far from being the best. Correct me if I am wrong, but
> abusing
> > of the logic I am using can heavily load the dialplan processing...
> > MFC's idea to execute an extension sounds pretty sexy.
> >
> > Can my example have undesirable side effects? Obviously the var name can
> be
> > something more unique.
> >
> > Cheers
> > On Sep 19, 2012, at 4:19 PM, Emrah <lists at kavun.ch> wrote:
> >
> >> You are correct indeed.
> >> This was just to get an idea. If I want data from an external source,
> there
> >> are plenty of dialplan options, including ${api func()} to be a little
> >> suicidal. :P
> >> We can pretty much fall back on LUA, Perl, IVR, so on and so forth for
> >> anything and everything.
> >> For the purpose of my test case, my example seems to fit the bill and
> does
> >> the job.
> >>
> >> Cheers,
> >> Emrah
> >> On Sep 19, 2012, at 3:57 PM, Stanislav Sinyagin <ssinyagin at yahoo.com>
> wrote:
> >>
> >>> Emrah, I think an IVR menu would be much easier to handle and to
> maintain
> >>> than what you;re trying to do.
> >>> Or write a Lua script, then you're even more flexible in all the
> checkups.
> >>> You can also look up the PIN in some external database instead of
> having it
> >>> statically coded in XML
> >>>
> >>>
> >>> From: Emrah <lists at kavun.ch>
> >>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >>> Sent: Wednesday, September 19, 2012 8:20 PM
> >>> Subject: Re: [Freeswitch-users] How to force FS to execute?
> >>>
> >>> There is a valid case to me, and here it is Sir. :)
> >>> <extension name="pin test">
> >>> <condition field="${pin}" expression="^1234$" break="on-true">
> >>> <action application="log" data="INFO Pin is ${pin}" />
> >>> <action application="transfer" data="3000 XML local" />
> >>> </condition>
> >>> <condition field="${pin}" expression="^\d{4}" break="never">
> >>> <action application="playback" data="conference/conf-bad-pin.wav" />
> >>> <action application="unset" data="pin" />
> >>> </condition>
> >>> <condition field="destination_number" expression="^111$" break="never">
> >>> <action application="answer"/>
> >>> <action application="sleep" data="1000"/>
> >>> <action application="read" data="4 4 conference/conf-pin.wav pin 10000
> #,*"
> >>> />
> >>> <action application="transfer" data="111 XML local" />
> >>> </condition>
> >>> </extension>
> >>>
> >>> I think I'm starting to get the hang of this thing!
> >>>
> >>> E
> >>>
> >>> On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>>
> >>>> There is no valid case not to use transfer (or even
> execute_extension) in
> >>>> this scenario. Create a new file: conf/dialplan/check_pin.xml:
> >>>>
> >>>> <include>
> >>>> <context name="utility">
> >>>>   <extension name="Check PIN">
> >>>>     <condition field="destination_number" expression="^check_pin$"/>
> >>>>     <condition field="${pin}" expression="^1234$">
> >>>>       <action application="transfer" data ="3000 XML local" />
> >>>>       <anti-action application="hangup" data="Login Failed"/>
> >>>>     </condition>
> >>>>   </extension>
> >>>> </context>
> >>>> </include>
> >>>>
> >>>> Then just do this in your original extension that has the read app
> (note I
> >>>> changed the name of the extension for clarity):
> >>>>
> >>>> <extension name="Get PIN from caller">
> >>>> <condition field="destination_number" expression="^111$"
> break="never">
> >>>>   <action application="answer"/>
> >>>>   <action application="sleep" data="1000"/>
> >>>>   <action application="read" data="4 4
> >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin
> 10000
> >>>> #,*"/>
> >>>>   <action application="log" data="INFO user entered '${pin}'"/>
> >>>>   <action application="transfer" data="check_pin XML utility"/>
> >>>> </condition>
> >>>> </extension>
> >>>>
> >>>> That's all there is to it. If this method seems weird, or unusual, or
> >>>> abstract then too bad! :) It's a great way to keep things relatively
> simple
> >>>> while ensuring a level of security.
> >>>>
> >>>> -MC
> >>>>
> >>>> On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
> >>>> Naming an extension won't help for my transfer purpose.
> >>>> What I think  I will do is use the same 1 extension idea, but verify
> the
> >>>> existance of ${pin} before executing the read instruction.
> >>>> This way if ${pin} exists, it will execute one set of instructions,
> if it
> >>>> doesn't, it will execute the read app. Following the read app would
> just be
> >>>> an action to transfer the user to the same context and
> destination_number.
> >>>>
> >>>> I hope what I tried to explain makes sense.
> >>>> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
> >>>>
> >>>>> You can call an extension whatever you want... It doesn't have to be
> a
> >>>>> number... But using the separate context allows you to keep the
> >>>>> destination
> >>>>> number field in the CDR something same like that actual destination
> >>>>> number...
> >>>>>
> >>>>>
> >>>>> On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
> >>>>>
> >>>>>> Is there a way to address an extension without assigning it to a
> >>>>>> destination_number? I don't want the extension to be reachable
> directly
> >>>>>> and
> >>>>>> would gladly avoid creating a dedicated context.
> >>>>>> Can I transfer a call to an extension that doesn't have a
> >>>>>> destination_number?
> >>>>>>
> >>>>>> Best,
> >>>>>> Emrah
> >>>>>> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
> >>>>>>
> >>>>>>> Easy indeed, but I wanted to avoid it thinking there was some other
> >>>>>>> alternative.
> >>>>>>> Thanks for your quick response, will use it.
> >>>>>>>
> >>>>>>> Cheers,
> >>>>>>> Emrah
> >>>>>>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org>
> wrote:
> >>>>>>>
> >>>>>>>> This is easy to handle... Do it as 2 different extensions and
> after you
> >>>>>>>> collect the PIN, use the transfer application to transfer the
> call to a
> >>>>>>>> pin
> >>>>>>>> validating extension... Problem fixed with a very simple solution
> >>>>>>>>
> >>>>>>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
> >>>>>>>>
> >>>>>>>>> Please bypass the inline="true"? A fantasy of mine.
> >>>>>>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
> >>>>>>>>>
> >>>>>>>>>> Hi all,
> >>>>>>>>>>
> >>>>>>>>>> My example below cannot work because FS does not store the
> value of
> >>>>>>>>>> ${pin}
> >>>>>>>>>> until the call goes into execution state. While keeping it
> simple,
> >>>>>>>>>> what
> >>>>>>>>>> would
> >>>>>>>>>> be your take on this? Is there a way to instruct FS to start
> >>>>>>>>>> executing at
> >>>>>>>>>> some point and discovering the dialplan along the way?
> >>>>>>>>>>
> >>>>>>>>>> <extension name="pin test">
> >>>>>>>>>> <condition field="destination_number" expression="^111$"
> >>>>>>>>>> break="never">
> >>>>>>>>>> <action application="answer"/>
> >>>>>>>>>> <action application="sleep" data="1000"/>
> >>>>>>>>>> <action application="read" data="4 4
> >>>>>>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav
> pin
> >>>>>>>>>> 10000
> >>>>>>>>>> #,*"
> >>>>>>>>>> inline="true" />
> >>>>>>>>>> </condition>
> >>>>>>>>>> <condition field="${pin}" expression="^1234$">
> >>>>>>>>>> <action application="transfer" data ="3000 XML local" />
> >>>>>>>>>> </condition>
> >>>>>>>>>> <condition field="${pin}" expression="^\d{4}$">
> >>>>>>>>>> <action application="hangup" />
> >>>>>>>>>> </condition>
> >>>>>>>>>> </extension>
> >>>>>>>>>>
> >>>>>>>>>> Thanks and all the best, as always.
> >>>>>>>>>> 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-use
> >>>>>>>>> rs
> >>>>>>>>> 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-user
> >>>>>>>>
> s
> >>>>>>>> http://www.freeswitch.org
> >>>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at 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
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at 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
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>



-- 
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 Sep 20 03:15:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Sep 2012 16:15:32 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call - What Would You Like
	To Talk About?
Message-ID: <CAKzWOxUvN-X_E+xSf_GSCVL3WH4n-_T6RUXF1n2oaWHgQVM+hw@mail.gmail.com>

Hey all,

While I'm thinking about it I would like to ask you to give me some ideas
on what we can discuss in our upcoming weekly conference calls. We have
openings from early October through the end of the year. I have a few more
lined up but I definitely would like to see some input from the community
on what else we can do on Wednesdays.

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From brian at freeswitch.org  Thu Sep 20 07:03:37 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Sep 2012 22:03:37 -0500
Subject: [Freeswitch-users] Demo IVR
Message-ID: <FA4B05B3-809F-4AC6-BFA4-AF0867B77F68@freeswitch.org>

This does actually ring a real person... aka ME.  Please don't tell me 'fuck you' when you call.  While I appreciate that... Its not fulfilling for me.

This is setup as a test for people to call, I do like talking to people from around the world, its fun.

Thanks,
--
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




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From curriegrad2004 at gmail.com  Thu Sep 20 07:54:54 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 19 Sep 2012 20:54:54 -0700
Subject: [Freeswitch-users] Demo IVR
In-Reply-To: <FA4B05B3-809F-4AC6-BFA4-AF0867B77F68@freeswitch.org>
References: <FA4B05B3-809F-4AC6-BFA4-AF0867B77F68@freeswitch.org>
Message-ID: <CAMWiMy8UM_T5J3QeVtP68ccZzkNmSH-F1W7ay769=cGHxyoamA@mail.gmail.com>

I can't stop laughing to myself when I read this on the IRC channel.
Maybe we should REALLY remove the option that lets the caller call
you.

On Wed, Sep 19, 2012 at 8:03 PM, Brian West <brian at freeswitch.org> wrote:
> This does actually ring a real person... aka ME.  Please don't tell me 'fuck
> you' when you call.  While I appreciate that... Its not fulfilling for me.
>
> This is setup as a test for people to call, I do like talking to people from
> around the world, its fun.
>
> Thanks,
> --
> 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
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ssinyagin at yahoo.com  Thu Sep 20 10:54:30 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 19 Sep 2012 23:54:30 -0700 (PDT)
Subject: [Freeswitch-users] Demo IVR
In-Reply-To: <CAMWiMy8UM_T5J3QeVtP68ccZzkNmSH-F1W7ay769=cGHxyoamA@mail.gmail.com>
References: <FA4B05B3-809F-4AC6-BFA4-AF0867B77F68@freeswitch.org>
	<CAMWiMy8UM_T5J3QeVtP68ccZzkNmSH-F1W7ay769=cGHxyoamA@mail.gmail.com>
Message-ID: <1348124070.63848.YahooMailNeo@web39306.mail.mud.yahoo.com>

I once got into the FreeSWITCH conference just by looking at the demo and dialing numbers. Was pretty surprised what those people were doing on my box :)

Tobi Oetiker, the author of RRDtool and MRTG and other cool things, also complains that people reach him for all kinds of IT questions just because his name is written on the front page of the tool.




>________________________________
> From: curriegrad2004 <curriegrad2004 at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, September 20, 2012 5:54 AM
>Subject: Re: [Freeswitch-users] Demo IVR
> 
>I can't stop laughing to myself when I read this on the IRC channel.
>Maybe we should REALLY remove the option that lets the caller call
>you.
>
>On Wed, Sep 19, 2012 at 8:03 PM, Brian West <brian at freeswitch.org> wrote:
>> This does actually ring a real person... aka ME.? Please don't tell me 'fuck
>> you' when you call.? While I appreciate that... Its not fulfilling for me.
>>
>> This is setup as a test for people to call, I do like talking to people from
>> around the world, its fun.
>>
>> Thanks,
>> --
>> 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
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
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From gavin.henry at gmail.com  Thu Sep 20 13:14:28 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 20 Sep 2012 10:14:28 +0100
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CAKzWOxVPNwFhBN6aUe7QQSyZ2qg4YN55e_5YquPUqkii-9Q6pQ@mail.gmail.com>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
	<CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
	<CAKzWOxVPNwFhBN6aUe7QQSyZ2qg4YN55e_5YquPUqkii-9Q6pQ@mail.gmail.com>
Message-ID: <CAA8_NKCY6XLZoSiSPuGWzz3aRrda5ZE3d+pgsdR0gDnfbKwevA@mail.gmail.com>

On 19 September 2012 23:47, Michael Collins <msc at freeswitch.org> wrote:
> I'm okay with the pledge list but I think GMV is giving Campbell a
> "please-don't-make-this-public" price because it is relatively low for a
> full sound set. (They love the FS project, so they do cool stuff like this.)
>
> If you want to help out please send a pledge message directly to Campbell
> Steven so that he can see what the interest is. Additionally, OSTAG would be
> happy to act as an agent and hold any donated funds in escrow until it's
> time to pay GMV. If you have money and are ready to donate now then simply
> Paypal money to donations at ostag.org and add instructions to seller saying
> "UK English sounds" and we'll know what that means. If you prefer to donate
> via a different method then check out our how-to-donate page.

Thanks. I'd like a quote first :-)

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


From gabe at gundy.org  Thu Sep 20 13:37:07 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Sep 2012 03:37:07 -0600
Subject: [Freeswitch-users] The subscriber appears to absent,
	but only sometimes and at some locations.
Message-ID: <CAMwZ-tKTUQOckuruMyUk5cdLZW4GQ36s=PJqb9DYXUezi3W=Ug@mail.gmail.com>

All,

I have a setup (happens to be domain base multi-tenancy) where we
occasionally have an issue where 1 or 2 (sometimes more) of the
extensions don't ring.

*******************************************************************************************************
LOG: Can't find user [some_user at location.main-domain.com]
LOG: Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
*******************************************************************************************************

Our bridge isn't anything special, but I put it out here so that
there's no question about what we're doing:

bridge(
[execute_on_originate=set_user::user1 at location.main-domain.com]user/user1 at location.main-domain.com,
[execute_on_originate=set_user::user2 at location.main-domain.com]user/user2 at location.main-domain.com,
[execute_on_originate=set_user::user3 at location.main-domain.com]user/user3 at location.main-domain.com,
)

Anyway, some locations report that not all extensions are ringing.
There doesn't seem to be a pattern about when it occurs or what
extension it happens to. While it seems to be only a hand full of
them, I suspect that it's under reported.

I know the extensions are all registered and I've tried calling each
directly and they work.

Anyone have any ideas?


Thanks,
Gabe


From ben at langfeld.co.uk  Thu Sep 20 13:42:44 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Thu, 20 Sep 2012 10:42:44 +0100
Subject: [Freeswitch-users] How to force FS to execute?
In-Reply-To: <CAKzWOxXc6+UFSSNXm25Vc-RC1+h0tD4B8iQY-HmjXriOeq0B7A@mail.gmail.com>
References: <72162FFC-A9F0-45E3-926D-00FDBF6004E7@kavun.ch>
	<CC7F9F8B.4D421%krice@freeswitch.org>
	<CAKzWOxXc6+UFSSNXm25Vc-RC1+h0tD4B8iQY-HmjXriOeq0B7A@mail.gmail.com>
Message-ID: <CAAyX+KHkfbum8RWVWZ7YF6kfLXWy0H3GiihvpKfsha-zt+2NXw@mail.gmail.com>

This sort of thing is trivial with Adhearsion:

class PINValidationController < Adhearsion::CallController
  def run
    menu 'conf-pin.wav', timeout: 10, terminator: '#' do
      match('1234') { speak 'Opening the pod bay doors' }

      timeout { speak 'You didn't enter anything!' }
      invalid do
        speak 'That's the wrong PIN'
        hangup
      end
    end
  end
end

Regards,
Ben Langfeld


On 19 September 2012 23:51, Michael Collins <msc at freeswitch.org> wrote:

> Don't we have a PIN validation routine in mod_voicemail? Alternatively,
> one could write a semi-generic PIN validation Lua script relatively easily.
> If no one does it prior to me getting to chapter 7 of the FS book re-write
> then I will consider doing it as an example script.
>
> -MC
>
>
> On Wed, Sep 19, 2012 at 2:12 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> If its extremely heavily loaded why not write an application to handle the
>> pin collection and validation
>>
>> K
>>
>>
>> On 9/19/12 3:31 PM, "Emrah" <lists at kavun.ch> wrote:
>>
>> > My example is far from being the best. Correct me if I am wrong, but
>> abusing
>> > of the logic I am using can heavily load the dialplan processing...
>> > MFC's idea to execute an extension sounds pretty sexy.
>> >
>> > Can my example have undesirable side effects? Obviously the var name
>> can be
>> > something more unique.
>> >
>> > Cheers
>> > On Sep 19, 2012, at 4:19 PM, Emrah <lists at kavun.ch> wrote:
>> >
>> >> You are correct indeed.
>> >> This was just to get an idea. If I want data from an external source,
>> there
>> >> are plenty of dialplan options, including ${api func()} to be a little
>> >> suicidal. :P
>> >> We can pretty much fall back on LUA, Perl, IVR, so on and so forth for
>> >> anything and everything.
>> >> For the purpose of my test case, my example seems to fit the bill and
>> does
>> >> the job.
>> >>
>> >> Cheers,
>> >> Emrah
>> >> On Sep 19, 2012, at 3:57 PM, Stanislav Sinyagin <ssinyagin at yahoo.com>
>> wrote:
>> >>
>> >>> Emrah, I think an IVR menu would be much easier to handle and to
>> maintain
>> >>> than what you;re trying to do.
>> >>> Or write a Lua script, then you're even more flexible in all the
>> checkups.
>> >>> You can also look up the PIN in some external database instead of
>> having it
>> >>> statically coded in XML
>> >>>
>> >>>
>> >>> From: Emrah <lists at kavun.ch>
>> >>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> >>> Sent: Wednesday, September 19, 2012 8:20 PM
>> >>> Subject: Re: [Freeswitch-users] How to force FS to execute?
>> >>>
>> >>> There is a valid case to me, and here it is Sir. :)
>> >>> <extension name="pin test">
>> >>> <condition field="${pin}" expression="^1234$" break="on-true">
>> >>> <action application="log" data="INFO Pin is ${pin}" />
>> >>> <action application="transfer" data="3000 XML local" />
>> >>> </condition>
>> >>> <condition field="${pin}" expression="^\d{4}" break="never">
>> >>> <action application="playback" data="conference/conf-bad-pin.wav" />
>> >>> <action application="unset" data="pin" />
>> >>> </condition>
>> >>> <condition field="destination_number" expression="^111$"
>> break="never">
>> >>> <action application="answer"/>
>> >>> <action application="sleep" data="1000"/>
>> >>> <action application="read" data="4 4 conference/conf-pin.wav pin
>> 10000 #,*"
>> >>> />
>> >>> <action application="transfer" data="111 XML local" />
>> >>> </condition>
>> >>> </extension>
>> >>>
>> >>> I think I'm starting to get the hang of this thing!
>> >>>
>> >>> E
>> >>>
>> >>> On Sep 19, 2012, at 12:22 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> >>>
>> >>>> There is no valid case not to use transfer (or even
>> execute_extension) in
>> >>>> this scenario. Create a new file: conf/dialplan/check_pin.xml:
>> >>>>
>> >>>> <include>
>> >>>> <context name="utility">
>> >>>>   <extension name="Check PIN">
>> >>>>     <condition field="destination_number" expression="^check_pin$"/>
>> >>>>     <condition field="${pin}" expression="^1234$">
>> >>>>       <action application="transfer" data ="3000 XML local" />
>> >>>>       <anti-action application="hangup" data="Login Failed"/>
>> >>>>     </condition>
>> >>>>   </extension>
>> >>>> </context>
>> >>>> </include>
>> >>>>
>> >>>> Then just do this in your original extension that has the read app
>> (note I
>> >>>> changed the name of the extension for clarity):
>> >>>>
>> >>>> <extension name="Get PIN from caller">
>> >>>> <condition field="destination_number" expression="^111$"
>> break="never">
>> >>>>   <action application="answer"/>
>> >>>>   <action application="sleep" data="1000"/>
>> >>>>   <action application="read" data="4 4
>> >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin
>> 10000
>> >>>> #,*"/>
>> >>>>   <action application="log" data="INFO user entered '${pin}'"/>
>> >>>>   <action application="transfer" data="check_pin XML utility"/>
>> >>>> </condition>
>> >>>> </extension>
>> >>>>
>> >>>> That's all there is to it. If this method seems weird, or unusual, or
>> >>>> abstract then too bad! :) It's a great way to keep things relatively
>> simple
>> >>>> while ensuring a level of security.
>> >>>>
>> >>>> -MC
>> >>>>
>> >>>> On Wed, Sep 19, 2012 at 8:46 AM, Emrah <lists at kavun.ch> wrote:
>> >>>> Naming an extension won't help for my transfer purpose.
>> >>>> What I think  I will do is use the same 1 extension idea, but verify
>> the
>> >>>> existance of ${pin} before executing the read instruction.
>> >>>> This way if ${pin} exists, it will execute one set of instructions,
>> if it
>> >>>> doesn't, it will execute the read app. Following the read app would
>> just be
>> >>>> an action to transfer the user to the same context and
>> destination_number.
>> >>>>
>> >>>> I hope what I tried to explain makes sense.
>> >>>> On Sep 19, 2012, at 10:40 AM, Ken Rice <krice at freeswitch.org> wrote:
>> >>>>
>> >>>>> You can call an extension whatever you want... It doesn't have to
>> be a
>> >>>>> number... But using the separate context allows you to keep the
>> >>>>> destination
>> >>>>> number field in the CDR something same like that actual destination
>> >>>>> number...
>> >>>>>
>> >>>>>
>> >>>>> On 9/19/12 7:25 AM, "Emrah" <lists at kavun.ch> wrote:
>> >>>>>
>> >>>>>> Is there a way to address an extension without assigning it to a
>> >>>>>> destination_number? I don't want the extension to be reachable
>> directly
>> >>>>>> and
>> >>>>>> would gladly avoid creating a dedicated context.
>> >>>>>> Can I transfer a call to an extension that doesn't have a
>> >>>>>> destination_number?
>> >>>>>>
>> >>>>>> Best,
>> >>>>>> Emrah
>> >>>>>> On Sep 19, 2012, at 7:10 AM, Emrah <lists at kavun.ch> wrote:
>> >>>>>>
>> >>>>>>> Easy indeed, but I wanted to avoid it thinking there was some
>> other
>> >>>>>>> alternative.
>> >>>>>>> Thanks for your quick response, will use it.
>> >>>>>>>
>> >>>>>>> Cheers,
>> >>>>>>> Emrah
>> >>>>>>> On Sep 19, 2012, at 12:50 AM, Ken Rice <krice at freeswitch.org>
>> wrote:
>> >>>>>>>
>> >>>>>>>> This is easy to handle... Do it as 2 different extensions and
>> after you
>> >>>>>>>> collect the PIN, use the transfer application to transfer the
>> call to a
>> >>>>>>>> pin
>> >>>>>>>> validating extension... Problem fixed with a very simple solution
>> >>>>>>>>
>> >>>>>>>> On 9/18/12 11:42 PM, "Emrah" <lists at kavun.ch> wrote:
>> >>>>>>>>
>> >>>>>>>>> Please bypass the inline="true"? A fantasy of mine.
>> >>>>>>>>> On Sep 19, 2012, at 12:32 AM, Emrah <lists at kavun.ch> wrote:
>> >>>>>>>>>
>> >>>>>>>>>> Hi all,
>> >>>>>>>>>>
>> >>>>>>>>>> My example below cannot work because FS does not store the
>> value of
>> >>>>>>>>>> ${pin}
>> >>>>>>>>>> until the call goes into execution state. While keeping it
>> simple,
>> >>>>>>>>>> what
>> >>>>>>>>>> would
>> >>>>>>>>>> be your take on this? Is there a way to instruct FS to start
>> >>>>>>>>>> executing at
>> >>>>>>>>>> some point and discovering the dialplan along the way?
>> >>>>>>>>>>
>> >>>>>>>>>> <extension name="pin test">
>> >>>>>>>>>> <condition field="destination_number" expression="^111$"
>> >>>>>>>>>> break="never">
>> >>>>>>>>>> <action application="answer"/>
>> >>>>>>>>>> <action application="sleep" data="1000"/>
>> >>>>>>>>>> <action application="read" data="4 4
>> >>>>>>>>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav
>> pin
>> >>>>>>>>>> 10000
>> >>>>>>>>>> #,*"
>> >>>>>>>>>> inline="true" />
>> >>>>>>>>>> </condition>
>> >>>>>>>>>> <condition field="${pin}" expression="^1234$">
>> >>>>>>>>>> <action application="transfer" data ="3000 XML local" />
>> >>>>>>>>>> </condition>
>> >>>>>>>>>> <condition field="${pin}" expression="^\d{4}$">
>> >>>>>>>>>> <action application="hangup" />
>> >>>>>>>>>> </condition>
>> >>>>>>>>>> </extension>
>> >>>>>>>>>>
>> >>>>>>>>>> Thanks and all the best, as always.
>> >>>>>>>>>> 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-use
>> >>>>>>>>> rs
>> >>>>>>>>> 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-user
>> >>>>>>>>
>> s
>> >>>>>>>> http://www.freeswitch.org
>> >>>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> _________________________________________________________________________
>> >>>>>> Professional FreeSWITCH Consulting Services:
>> >>>>>> consulting at freeswitch.org
>> >>>>>> http://www.freeswitchsolutions.com
>> >>>>>>
>> >>>>>> 
>> >>>>>> 
>> >>>>>>
>> >>>>>> Official FreeSWITCH Sites
>> >>>>>> http://www.freeswitch.org
>> >>>>>> http://wiki.freeswitch.org
>> >>>>>> http://www.cluecon.com
>> >>>>>>
>> >>>>>> FreeSWITCH-users mailing list
>> >>>>>> FreeSWITCH-users at 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
>> >>>>
>> >>>>
>> >>>>
>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> 
>> >>>> 
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://wiki.freeswitch.org
>> >>>> http://www.cluecon.com
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at 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
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>
>
>
> --
> 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 me at nevian.org  Thu Sep 20 14:10:59 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 20 Sep 2012 14:10:59 +0400
Subject: [Freeswitch-users] Case insensitive SIP URI
Message-ID: <1205261348135859@web2h.yandex.ru>

Hello,

We have multiple registrations turned on on both profiles and few of our users registered at Internal, some on External.
Via sofia_conntact we are able to reach them all BUT some registers for example as Support at domain.tld and others as support at domain.tld - and this is big problem for us :(

I believe it's incorrect behavior to match case sensitive although it's "by design" (as per Jira).

Anyone can propose workaround or patch for Sofia?

Thanx in advance.

-- 
wbr,
Serge


From brian at freeswitch.org  Thu Sep 20 17:07:50 2012
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Sep 2012 08:07:50 -0500
Subject: [Freeswitch-users] Demo IVR
In-Reply-To: <CAMWiMy8UM_T5J3QeVtP68ccZzkNmSH-F1W7ay769=cGHxyoamA@mail.gmail.com>
References: <FA4B05B3-809F-4AC6-BFA4-AF0867B77F68@freeswitch.org>
	<CAMWiMy8UM_T5J3QeVtP68ccZzkNmSH-F1W7ay769=cGHxyoamA@mail.gmail.com>
Message-ID: <1FC1A4B7-8927-4A8D-9E2F-85FC8EBAA773@freeswitch.org>

LOL, I don't mind it when people talk but when they are rude about it and call over and over and hangup on purpose... I actually meet a lot of folks via that.
On Sep 19, 2012, at 10:54 PM, curriegrad2004 wrote:

> I can't stop laughing to myself when I read this on the IRC channel.
> Maybe we should REALLY remove the option that lets the caller call
> you.

--
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




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From trob at freemail.hu  Thu Sep 20 17:40:44 2012
From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=)
Date: Thu, 20 Sep 2012 15:40:44 +0200
Subject: [Freeswitch-users] recording's channels shifted
Message-ID: <505B1CDC.40408@freemail.hu>

Hi

Why left and right channels' voice shifted from each other in the recorded mp3?

I record the sessions' voice thios way:
<action application="set" data="execute_on_answer=record_session 
$${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/>
When i listen back the mp3, at begining the two chennels are synchonized, but after some minutes 
there will be a shifting between the channels.

Why?

thanks,
Robert
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From dvl36.ripe.nick at gmail.com  Thu Sep 20 18:05:09 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Thu, 20 Sep 2012 17:05:09 +0300
Subject: [Freeswitch-users] Case insensitive SIP URI
In-Reply-To: <1205261348135859@web2h.yandex.ru>
References: <1205261348135859@web2h.yandex.ru>
Message-ID: <CAMjaK3C=E7eDd96+cGGqwMRpack8Whr25ag+kAJSJhy3CNFG6w@mail.gmail.com>

Hello, Serge

2012/9/20 Serge S. Yuriev <me at nevian.org>

> I believe it's incorrect behavior to match case sensitive although it's
> "by design" (as per Jira).
>

According to rfc3261 <http://www.rfc-editor.org/rfc/rfc3261.txt> username
part of SIP URI must be case sensitive.
BTW this is common case in internet&unix world.

WBR,
 Dmitry.
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From me at nevian.org  Thu Sep 20 20:39:21 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 20 Sep 2012 20:39:21 +0400
Subject: [Freeswitch-users] Case insensitive SIP URI
In-Reply-To: <CAMjaK3C=E7eDd96+cGGqwMRpack8Whr25ag+kAJSJhy3CNFG6w@mail.gmail.com>
References: <1205261348135859@web2h.yandex.ru>
	<CAMjaK3C=E7eDd96+cGGqwMRpack8Whr25ag+kAJSJhy3CNFG6w@mail.gmail.com>
Message-ID: <277921348159161@web24g.yandex.ru>

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From avi at avimarcus.net  Thu Sep 20 20:47:46 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 20 Sep 2012 19:47:46 +0300
Subject: [Freeswitch-users] Case insensitive SIP URI
In-Reply-To: <277921348159161@web24g.yandex.ru>
References: <1205261348135859@web2h.yandex.ru>
	<CAMjaK3C=E7eDd96+cGGqwMRpack8Whr25ag+kAJSJhy3CNFG6w@mail.gmail.com>
	<277921348159161@web24g.yandex.ru>
Message-ID: <CALNZuwnH2T+q=Upo7rz5ZdyS=LTN20kB42ay-2q+Sa3yXxkXCg@mail.gmail.com>

Have the boxes register with the same case username.
Or have everyone register to numerical extensions, those don't have cases...

-Avi


On Thu, Sep 20, 2012 at 7:39 PM, Serge S. Yuriev <me at nevian.org> wrote:

> Hello,
>
> 20.09.2012, 18:05, "Dmitry Lysenko" <dvl36.ripe.nick at gmail.com>:
>
> Hello, Serge
>
> 2012/9/20 Serge S. Yuriev <me at nevian.org>
>
> I believe it's incorrect behavior to match case sensitive although it's
> "by design" (as per Jira).
>
>
> According to rfc3261 <http://www.rfc-editor.org/rfc/rfc3261.txt> username
> part of SIP URI must be case sensitive.
> BTW this is common case in internet&unix world.
>
>
>
> Thank you for pointing out!
> But I'm a bit puzzled - what for this can be needed??
>
> And once more - any suggestions on workaround?
> --
> wbr,
> Serge
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Sep 20 21:14:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Sep 2012 10:14:41 -0700
Subject: [Freeswitch-users] recording's channels shifted
In-Reply-To: <505B1CDC.40408@freemail.hu>
References: <505B1CDC.40408@freemail.hu>
Message-ID: <CAKzWOxX289MUOLV2eDbiFHz5mYjJTtnLF-Wgp7GgVooodYDiYQ@mail.gmail.com>

This is more likely an issue with the media streams than the recording
itself. Do the callers notice anything? Is one or both callers outside the
local network where the FS server reside? Just looking for places where
interesting/weird things might happen.

-MC

On Thu, Sep 20, 2012 at 6:40 AM, T?th R?bert <trob at freemail.hu> wrote:

>  Hi
>
> Why left and right channels' voice shifted from each other in the recorded
> mp3?
>
> I record the sessions' voice thios way:
> <action application="set" data="execute_on_answer=record_session
> $${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/>
> When i listen back the mp3, at begining the two chennels are synchonized,
> but after some minutes there will be a shifting between the channels.
>
> Why?
>
> thanks,
> Robert
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gabe at gundy.org  Thu Sep 20 21:21:20 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Sep 2012 11:21:20 -0600
Subject: [Freeswitch-users] The subscriber appears to absent,
	but only sometimes and at some locations.
In-Reply-To: <CAMwZ-tKTUQOckuruMyUk5cdLZW4GQ36s=PJqb9DYXUezi3W=Ug@mail.gmail.com>
References: <CAMwZ-tKTUQOckuruMyUk5cdLZW4GQ36s=PJqb9DYXUezi3W=Ug@mail.gmail.com>
Message-ID: <CAMwZ-tJ3xwEUOuEH7xfJGL3kkuCdvqEDEpMoptR5EF2rLpLKkQ@mail.gmail.com>

On Thu, Sep 20, 2012 at 3:37 AM, Gabriel Gunderson <gabe at gundy.org> wrote:
> Anyway, some locations report that not all extensions are ringing.
> There doesn't seem to be a pattern about when it occurs or what
> extension it happens to. While it seems to be only a hand full of
> them, I suspect that it's under reported.

So, I want to be careful when reporting this (I hate to give bad
info), but it seems like this might happen more with server load. Of
course it might also be that because we have more calls coming in that
there are more chances to notice it.

Anyone have ideas on why the random SUBSCRIBER_ABSENT? Can it be as
simple as the firewall? And if so, why wouldn't it be constant?

Thanks,
Gabe


From lconroy at insensate.co.uk  Thu Sep 20 21:29:03 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Thu, 20 Sep 2012 18:29:03 +0100
Subject: [Freeswitch-users] Case insensitive SIP URI
In-Reply-To: <277921348159161@web24g.yandex.ru>
References: <1205261348135859@web2h.yandex.ru>
	<CAMjaK3C=E7eDd96+cGGqwMRpack8Whr25ag+kAJSJhy3CNFG6w@mail.gmail.com>
	<277921348159161@web24g.yandex.ru>
Message-ID: <3694A803-32DB-4F12-9669-7E58FD711A56@insensate.co.uk>

Hi there,
 this goes deep into SIP's early history.
SIP inherited its userpart at domainpart syntax from email. Email userparts are case sensitive.
Also, HTTP had and has case sensitivity as well.
Remember, these rules were being set in the late '90s.
IIRC, the arguments at the time concerned having to deal with case insensitive comparisons,
and the royal pain that would be introduced if we ever wanted to deal with non-ASCII IDs.

Just to confuse the innocent, the domainpart (effectively) isn't case sensitive because
it's going to be passed to DNS, and DNS Fully Qualified Domain Names are case INsensitive.

[OK ...a hidden case aware form is used in DNS queries with recent software to help block
 certain kinds of DNS attacks -- but that's entirely transparent to DNS users]

There has been sooooooo much water passing under that particular bridge it is not going
to change any time soon.

all the best,
  Lawrence


On 20 Sep 2012, at 17:39, Serge S. Yuriev wrote:
> Hello,
>  
> 20.09.2012, 18:05, "Dmitry Lysenko" <dvl36.ripe.nick at gmail.com>:
>> Hello, Serge
>> 
>> 2012/9/20 Serge S. Yuriev <me at nevian.org>
>> I believe it's incorrect behavior to match case sensitive although it's "by design" (as per Jira).
>>  
>> According to rfc3261 username part of SIP URI must be case sensitive.
>> BTW this is common case in internet&unix world.
>  
>  
> Thank you for pointing out!
> But I'm a bit puzzled - what for this can be needed??
>  
> And once more - any suggestions on workaround?
> -- 
> wbr,
> Serge
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dvl36.ripe.nick at gmail.com  Thu Sep 20 21:37:08 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Thu, 20 Sep 2012 20:37:08 +0300
Subject: [Freeswitch-users] Case insensitive SIP URI
In-Reply-To: <277921348159161@web24g.yandex.ru>
References: <1205261348135859@web2h.yandex.ru>
	<CAMjaK3C=E7eDd96+cGGqwMRpack8Whr25ag+kAJSJhy3CNFG6w@mail.gmail.com>
	<277921348159161@web24g.yandex.ru>
Message-ID: <CAMjaK3C_cBB3+PXvAhbbRjwDSZ5Db3-w9v6oBorFRQDV3mm1Vg@mail.gmail.com>

Hello
2012/9/20 Serge S. Yuriev <me at nevian.org>

> 20.09.2012, 18:05, "Dmitry Lysenko" <dvl36.ripe.nick at gmail.com>:
>
> According to rfc3261 <http://www.rfc-editor.org/rfc/rfc3261.txt> username
> part of SIP URI must be case sensitive.
> BTW this is common case in internet&unix world.
>
>
>
> Thank you for pointing out!
> But I'm a bit puzzled - what for this can be needed??
>

Main goal, IMO, is to simplify firmware of sip-phones, because
case-insensitivity is hard to internationalize.


> And once more - any suggestions on workaround?
>

Sofia sources hacking.
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From mthakershi at gmail.com  Thu Sep 20 21:56:15 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 20 Sep 2012 12:56:15 -0500
Subject: [Freeswitch-users] Speak vs. StreamFile
Message-ID: <CAKHK4ZPMn8G=7qByqgBePYVrB-2qXrZBG5ScrBATthQnAzJpgw@mail.gmail.com>

Hello,

I am working on optimizing call flow code. I use Cepstral as TTS from FS
managed module (C# .net code).

I suspect using Session.Speak is slowing down the call experience (prompt
delay) when there are # of concurrent calls.

My question: does it help with performance if I create WAV file for each
static prompt in advance and use Session.StreamFile instead of
Session.Speak?

Any help is appreciated.

Malay
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From avi at avimarcus.net  Thu Sep 20 22:00:11 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 20 Sep 2012 21:00:11 +0300
Subject: [Freeswitch-users] Speak vs. StreamFile
In-Reply-To: <CAKHK4ZPMn8G=7qByqgBePYVrB-2qXrZBG5ScrBATthQnAzJpgw@mail.gmail.com>
References: <CAKHK4ZPMn8G=7qByqgBePYVrB-2qXrZBG5ScrBATthQnAzJpgw@mail.gmail.com>
Message-ID: <CALNZuw=mhmxzBo73cTyaAho8PfwDReBAn6ZDgJwFu7VampJBRw@mail.gmail.com>

Give it a try... if you've got the volume, even better than wav is native
file: http://wiki.freeswitch.org/wiki/Mod_native_file
-Avi


On Thu, Sep 20, 2012 at 8:56 PM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Hello,
>
> I am working on optimizing call flow code. I use Cepstral as TTS from FS
> managed module (C# .net code).
>
> I suspect using Session.Speak is slowing down the call experience (prompt
> delay) when there are # of concurrent calls.
>
> My question: does it help with performance if I create WAV file for each
> static prompt in advance and use Session.StreamFile instead of
> Session.Speak?
>
> Any help is appreciated.
>
> Malay
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Sep 20 22:42:34 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 20 Sep 2012 11:42:34 -0700 (PDT)
Subject: [Freeswitch-users] Speak vs. StreamFile
In-Reply-To: <CALNZuw=mhmxzBo73cTyaAho8PfwDReBAn6ZDgJwFu7VampJBRw@mail.gmail.com>
References: <CAKHK4ZPMn8G=7qByqgBePYVrB-2qXrZBG5ScrBATthQnAzJpgw@mail.gmail.com>
	<CALNZuw=mhmxzBo73cTyaAho8PfwDReBAn6ZDgJwFu7VampJBRw@mail.gmail.com>
Message-ID: <1348166554.17472.YahooMailNeo@web39305.mail.mud.yahoo.com>

interesting. Does mod_native_file support transcoding when the desired encoding is not found on the disk?




>________________________________
> From: Avi Marcus <avi at avimarcus.net>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, September 20, 2012 8:00 PM
>Subject: Re: [Freeswitch-users] Speak vs. StreamFile
> 
>
>Give it a try... if you've got the volume, even better than wav is native file:?http://wiki.freeswitch.org/wiki/Mod_native_file
>
>-Avi
>
>
>On Thu, Sep 20, 2012 at 8:56 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
>
>Hello,
>>
>>
>>I am working on optimizing call flow code. I use Cepstral as TTS from FS managed module (C# .net code).
>>
>>
>>I suspect using Session.Speak is slowing down the call experience (prompt delay) when there are # of concurrent calls.
>>
>>
>>My question: does it help with performance if I create WAV file for each static prompt in advance and use Session.StreamFile instead of Session.Speak?
>>
>>
>>Any help is appreciated.
>>
>>Malay
>>
>>
>>
>>
>>
>
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From garmt.noname at gmail.com  Thu Sep 20 23:02:53 2012
From: garmt.noname at gmail.com (grmt)
Date: Thu, 20 Sep 2012 21:02:53 +0200
Subject: [Freeswitch-users] Speak vs. StreamFile
In-Reply-To: <CALNZuw=mhmxzBo73cTyaAho8PfwDReBAn6ZDgJwFu7VampJBRw@mail.gmail.com>
References: <CAKHK4ZPMn8G=7qByqgBePYVrB-2qXrZBG5ScrBATthQnAzJpgw@mail.gmail.com>
	<CALNZuw=mhmxzBo73cTyaAho8PfwDReBAn6ZDgJwFu7VampJBRw@mail.gmail.com>
Message-ID: <020f01cd9762$8a60a0b0$9f21e210$@gmail.com>

If you still want to use TTS, you may want to cache your generated TTS, e.g.
http://www.tomp.co.uk/blog/view/23

Note that your TTS license may not allow you to do so ...

 

Alternatively, you can gain some cpu cycles by setting cache_speech_handles
to true in your dialplan.

e.g

<extension name="use_tts">

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

  

               use tts

               use tts again

</extension>

 

Not sure how many cycles you will gain though.

 

Garmt

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Thursday, September 20, 2012 20:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Speak vs. StreamFile

 

Give it a try... if you've got the volume, even better than wav is native
file: http://wiki.freeswitch.org/wiki/Mod_native_file


-Avi

 

On Thu, Sep 20, 2012 at 8:56 PM, Malay Thakershi <mthakershi at gmail.com>
wrote:

Hello,

 

I am working on optimizing call flow code. I use Cepstral as TTS from FS
managed module (C# .net code).

 

I suspect using Session.Speak is slowing down the call experience (prompt
delay) when there are # of concurrent calls.

 

My question: does it help with performance if I create WAV file for each
static prompt in advance and use Session.StreamFile instead of
Session.Speak?

 

Any help is appreciated.

 

Malay

 

 


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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Sep 21 00:34:42 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 20 Sep 2012 21:34:42 +0100
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
	<CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
Message-ID: <CAMbW9o+PBYiX6SV1f0caokiw1wLQPagACB1PBen+4bx3_4hTXA@mail.gmail.com>

Strangely enough I was talking to the wife about trying this the other day.
Not professionally though, haven't got the funding.
On 19 Sep 2012 19:10, "Avi Marcus" <avi at avimarcus.net> wrote:

> Could we get a quote and a pledge list, to see how short we are?
>
> -Avi
>
>
> On Wed, Sep 19, 2012 at 8:59 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Hey Guys,
>>
>> One thing we could try, is to get OSTAG involved here... Get a quote from
>> GM Voices and then get several of you en_UK guys to donated a few quid to
>> OSTAG to cover the cost and OSTAG orders it and donates it opensource to
>> FreeSWITCH everyone wins
>>
>> K
>>
>>
>>
>> On 9/19/12 12:54 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:
>>
>> Hi Campbell,
>>
>> I run SureVoIP and would love to do this. We've tired to get a quote
>> before but didn't get anywhere. Happy for all to go to the project.
>>
>> Thanks.
>>
>> On Wednesday, 12 September 2012, Campbell Steven  wrote:
>>
>> Hi en_UK FreeSWITCH users,
>>
>> I'm trying to get a full prompt set (every prompt currently recorded
>> by Callie including the ztrp prompts etc..) organised. The talent
>> would be female and prompts would be recorded through GM Voices and
>> therefore easy to get matching custom recordings for your applications
>> into the future. The prompt set would then be given to the FreeSWITCH
>> project for distribution as per the existing prompts.
>>
>> Anyone who is willing to help me out with funding this or has any
>> questions please contact me off-list as I'd really like to get this
>> sorted. Every little bit counts here so even if you just want to drop
>> a few $ in it all adds up.
>>
>> Thanks
>>
>> Campbell
>>
>> _________________________________________________________________________
>> 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
>>
>>
>>
>> --
>> 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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Sep 21 00:53:24 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 20 Sep 2012 22:53:24 +0200
Subject: [Freeswitch-users] extra headers to event message to locate user
	directory for voicemail
Message-ID: <505B8244.6020403@gmail.com>

Hello,

I was trying to build user directory file dynamically via a lua script 
at a dial-in event to listen to/leave a voice message, looking to pass 
some values from particular SIP headers.

But the session object seems to be unavailable at that moment - next are 
the relevant config snippets, the dialplan:

     <extension name="vbox">
       <condition field="destination_number" expression="^vb-(.+)$">
         <action application="answer"/>
         <action application="voicemail" data="default ${domain_name} $1"/>
       </condition>
     </extension>

and lua.conf:

    <param name="xml-handler-script" 
value="/usr/local/freeswitch/scripts/dp.lua"/>
     <param name="xml-handler-bindings" value="directory"/>

The event message is like the one from:
http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration#when_being_called_.28by_another_extension.29

The Core-UUID is different that channel UUID, so I could not find any 
API commands that could be used to get channel variables.

Do I miss any setting/command that should get me access to channel 
variables or SIP headers inside the dp.lua?

My quick workaround was to patch to be able to add extra headers to the 
internal event message (most of the time diving in the sources is the 
fast way to get things done :-) ).

I am attaching the patch for the case there is no other solution and 
someone else is interested - by now it adds extra headers only when 
checking voicemail or recording a new message, but should be 
straightforward to add to other places across voicemail module if it has 
to (I can make the patch if one points other types of voicemail events 
that can benefit of extra headers).

The patch adds a new channel variable, locate_event_extra_headers, that 
can be set to a prefix of the channel variable names that have to be 
passed as extra headers in the internal event message fired to locate 
user directory. An example is in the patch comments.

If there is no solution to pass channel variables/SIP headers to the lua 
scripts used for generating user directories, would it be a good feature 
to add to devel branch? My particular need comes from using Kamailio as 
the SIP server and when user is offline, the call is directed to 
FreeSwitch for voicemail service. Also users can dial in to listed to 
messages. I want to pass voicemail box id, pincode and email via SIP 
headers, as I have them already loaded in Kamailio config script, 
avoiding another database lookup from freeswitch lua script.

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
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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>From 6aea6c27f78a7799b440a06be4c158f6f61bf601 Mon Sep 17 00:00:00 2001
From: Daniel-Constantin Mierla <miconda at gmail.com>
Date: Thu, 20 Sep 2012 22:05:12 +0200
Subject: [PATCH] mod_voicemail: allow passing extra headers in event message
 to locate user directory

- channel variable 'locate_event_extra_headers' can be set to the prefix
  of channel variable names that have to be added as extra headers to
  the event message fired to locate the user directory
- the header name is variable name and the header body is variable value
- here is an example of adding the value of SIP header X-Data as event
  message header X-Event-Data:

<action application="set" data="locate_event_extra_headers=X-Event-" />
<action application="set" data="X-Event-Data=${sip_h_X-Data}" />

- the headers are added for leave voice message and listen voice message
  events
- the extra headers can be useful on building user directory
  dinamically, e.g., via a Lua script
---
 src/mod/applications/mod_voicemail/mod_voicemail.c |   30 ++++++++++++++++++++
 1 file changed, 30 insertions(+)

diff --git a/src/mod/applications/mod_voicemail/mod_voicemail.c b/src/mod/applications/mod_voicemail/mod_voicemail.c
index 51677c5..9509ea7 100644
--- a/src/mod/applications/mod_voicemail/mod_voicemail.c
+++ b/src/mod/applications/mod_voicemail/mod_voicemail.c
@@ -161,6 +161,31 @@ struct vm_profile {
 typedef struct vm_profile vm_profile_t;
 
 
+static int voicemail_event_message_add_extra_headers(switch_channel_t *channel, switch_event_t *event,
+		const char *prefix)
+{
+    switch_event_header_t *hi = NULL;
+	int n;
+
+	if(channel==NULL || event==NULL || prefix==NULL)
+		return 0;
+
+	n = 0;
+    if ((hi = switch_channel_variable_first(channel))) {
+        for (; hi; hi = hi->next) {
+            const char *name = (char *) hi->name;
+            char *value = (char *) hi->value;
+
+            if (!strncasecmp(name, prefix, strlen(prefix))) {
+				switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, name, value);
+            }
+        }
+        switch_channel_variable_last(channel);
+    }
+
+    return n;
+}
+
 switch_cache_db_handle_t *vm_get_db_handle(vm_profile_t *profile)
 {
 	switch_cache_db_connection_options_t options = { {0} };
@@ -2326,6 +2351,8 @@ static void voicemail_check_main(switch_core_session_t *session, vm_profile_t *p
 
 					switch_event_create(&params, SWITCH_EVENT_GENERAL);
 					switch_assert(params);
+					voicemail_event_message_add_extra_headers(channel, params,
+							switch_channel_get_variable(channel, "locate_event_extra_headers"));
 					switch_event_add_header_string(params, SWITCH_STACK_BOTTOM, "action", "voicemail-lookup");
 					switch_event_add_header_string(params, SWITCH_STACK_BOTTOM, "destination_number", caller_profile->destination_number);
 					switch_event_add_header_string(params, SWITCH_STACK_BOTTOM, "caller_id_number", caller_id_number);
@@ -3157,6 +3184,7 @@ static switch_status_t voicemail_inject(const char *data, switch_core_session_t
 	return status;
 }
 
+
 static switch_status_t voicemail_leave_main(switch_core_session_t *session, vm_profile_t *profile, const char *domain_name, const char *id)
 {
 	switch_channel_t *channel = switch_core_session_get_channel(session);
@@ -3220,6 +3248,8 @@ static switch_status_t voicemail_leave_main(switch_core_session_t *session, vm_p
 		switch_event_create(&locate_params, SWITCH_EVENT_REQUEST_PARAMS);
 		switch_assert(locate_params);
 		switch_event_add_header_string(locate_params, SWITCH_STACK_BOTTOM, "action", "voicemail-lookup");
+		voicemail_event_message_add_extra_headers(channel, locate_params,
+					switch_channel_get_variable(channel, "locate_event_extra_headers"));
 
 		if (switch_xml_locate_user_merged("id", id, domain_name, switch_channel_get_variable(channel, "network_addr"),
 										  &x_user, locate_params) == SWITCH_STATUS_SUCCESS) {
-- 
1.7.9.2


From msc at freeswitch.org  Fri Sep 21 00:56:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Sep 2012 13:56:14 -0700
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CAMbW9o+PBYiX6SV1f0caokiw1wLQPagACB1PBen+4bx3_4hTXA@mail.gmail.com>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
	<CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
	<CAMbW9o+PBYiX6SV1f0caokiw1wLQPagACB1PBen+4bx3_4hTXA@mail.gmail.com>
Message-ID: <CAKzWOxU55GCJY7XLh0ECkwk==+Yan87VgYyx-m-O9nm0-ORyrA@mail.gmail.com>

Well, only one or two people have shown any interest at all and AFAIK we
haven't had any pledge any money, although Campbell can update us on that.
Keep in mind that you don't need to know the final price if you just want
to pledge $25 or whatever to the final cost. A lot of little contributions
will add up, so don't be shy!

-MC

On Thu, Sep 20, 2012 at 1:34 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

>
> Strangely enough I was talking to the wife about trying this the other
> day. Not professionally though, haven't got the funding.
> On 19 Sep 2012 19:10, "Avi Marcus" <avi at avimarcus.net> wrote:
>
>> Could we get a quote and a pledge list, to see how short we are?
>>
>> -Avi
>>
>>
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From trob at freemail.hu  Fri Sep 21 01:03:17 2012
From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=)
Date: Thu, 20 Sep 2012 23:03:17 +0200
Subject: [Freeswitch-users]  recording's channels shifted
Message-ID: <505B8495.5060601@freemail.hu>

No, the callers do not notice anything, they hear each other without problem.

One party is on the local network where FS is, and other is outside.
But as they do not notice this problem, i think the problem is at the recording. I think.


>/  /This is more likely an issue with the media streams than the recording
>/  /itself. Do the callers notice anything? Is one or both callers outside the
>/  /local network where the FS server reside? Just looking for places where
>/  /interesting/weird things might happen.
>/  /
>/  /-MC
>/  /
>/  /On Thu, Sep 20, 2012 at 6:40 AM, T?th R?bert <trob at freemail.hu  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
>/  /
>/  />/   Hi
/>/  />/
/>/  />/  Why left and right channels' voice shifted from each other in the recorded
/>/  />/  mp3?
/>/  />/
/>/  />/  I record the sessions' voice thios way:
/> >/  <action application="set" data="execute_on_answer=record_session
/> >/  $${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/>
/> >/  When i listen back the mp3, at begining the two chennels are synchonized,
/> >/  but after some minutes there will be a shifting between the channels.
/> >/
/> >/  Why?
/> >/
/> >/  thanks,
/> >/  Robert/

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From msc at freeswitch.org  Fri Sep 21 01:24:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Sep 2012 14:24:43 -0700
Subject: [Freeswitch-users] recording's channels shifted
In-Reply-To: <505B8495.5060601@freemail.hu>
References: <505B8495.5060601@freemail.hu>
Message-ID: <CAKzWOxX2VfzHyAmsTtDQo8MWNWUx3-4=WqNkVMoLSZm-yazu4g@mail.gmail.com>

I recommend that you get a pcap of such a conversation and analyze it in
Wireshark. See if the lag shows up in the analysis or not. That will at
least give you a clue as to where the issue is. (Compare the recording on
disk with the playback in Wireshark. Note: you'll need the call to be g.711
for this to work.)

-MC

On Thu, Sep 20, 2012 at 2:03 PM, T?th R?bert <trob at freemail.hu> wrote:

>  No, the callers do not notice anything, they hear each other without problem.
>
> One party is on the local network where FS is, and other is outside.
> But as they do not notice this problem, i think the problem is at the recording. I think.
>
>
> >* *This is more likely an issue with the media streams than the recording
> >* *itself. Do the callers notice anything? Is one or both callers outside the
> >* *local network where the FS server reside? Just looking for places where
> >* *interesting/weird things might happen.
> >* *
> >* *-MC
> >* *
> >* *On Thu, Sep 20, 2012 at 6:40 AM, T?th R?bert <trob at freemail.hu <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
> >* *
> >* *>*  Hi*>* *>**>* *>* Why left and right channels' voice shifted from each other in the recorded*>* *>* mp3?*>* *>**>* *>* I record the sessions' voice thios way:*> >* <action application="set" data="execute_on_answer=record_session*> >* $${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/>*> >* When i listen back the mp3, at begining the two chennels are synchonized,*> >* but after some minutes there will be a shifting between the channels.*> >**> >* Why?*> >**> >* thanks,*> >* Robert*
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 casteven at gmail.com  Fri Sep 21 01:50:39 2012
From: casteven at gmail.com (Campbell Steven)
Date: Fri, 21 Sep 2012 09:50:39 +1200
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CAKzWOxU55GCJY7XLh0ECkwk==+Yan87VgYyx-m-O9nm0-ORyrA@mail.gmail.com>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
	<CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
	<CAMbW9o+PBYiX6SV1f0caokiw1wLQPagACB1PBen+4bx3_4hTXA@mail.gmail.com>
	<CAKzWOxU55GCJY7XLh0ECkwk==+Yan87VgYyx-m-O9nm0-ORyrA@mail.gmail.com>
Message-ID: <CAPerZE9uebr-hGnO-YzKoR-qYxAP9UPVCeO7+4BVJL+4Nbz=jQ@mail.gmail.com>

Hi,

So far I have 3 people (from this thread) that have shown interest, if
I include myself then we have 4, I will respond offlist to those who
are interested in going further and see what we can come up with.

I have gone down the path of holding a private prompt set before, and
it's a laborious and difficult project I don't wish to do again, and
want to help the rest of the community avoid it. With GM Voices
everyone can get their own custom prompts with matching talent going
forward.

Campbell

On 21 September 2012 08:56, Michael Collins <msc at freeswitch.org> wrote:
> Well, only one or two people have shown any interest at all and AFAIK we
> haven't had any pledge any money, although Campbell can update us on that.
> Keep in mind that you don't need to know the final price if you just want to
> pledge $25 or whatever to the final cost. A lot of little contributions will
> add up, so don't be shy!
>
> -MC
>
>
> On Thu, Sep 20, 2012 at 1:34 PM, Andrew Cassidy
> <andrew at cassidywebservices.co.uk> wrote:
>>
>>
>> Strangely enough I was talking to the wife about trying this the other
>> day. Not professionally though, haven't got the funding.
>>
>> On 19 Sep 2012 19:10, "Avi Marcus" <avi at avimarcus.net> wrote:
>>>
>>> Could we get a quote and a pledge list, to see how short we are?
>>>
>>> -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 ljjimenez at gmail.com  Fri Sep 21 01:59:39 2012
From: ljjimenez at gmail.com (ljjimenez at gmail.com)
Date: Thu, 20 Sep 2012 17:59:39 -0400
Subject: [Freeswitch-users] Convert to e.164
Message-ID: <F36E5CE5-2F07-4FB4-B327-46B99E51277A@gmail.com>


Hello list, I have a provider who needs to receive the call prefixed with a "+" in the "From" and "To" headers, is there any dial plan function for this or how is the best practice to modify the "From" and "To" headers to prefix the + ?

Thanks in advance

Luis Jim?nez 
UDR



From shaheryarkh at googlemail.com  Fri Sep 21 02:04:00 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 21 Sep 2012 00:04:00 +0200
Subject: [Freeswitch-users] Segmentation Fault while running configure
Message-ID: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>

Hi,

I am running Ubuntu 12.04 64-bit in KVM and getting segmentation fault
while running configure script of latest stable Freeswitch v1.2. Here are
lines just before crash.

/configure --enable-64 --enable-core-odbc-support --enable-optimization
--with-pic
...
checking whether getpwnam_r and getpwuid_r are posix like... yes
checking whether getpwnam_r and getpwuid_r are posix _draft_ like... no
configure: creating ./config.status
./configure.gnu: line 3: 20996 Segmentation fault      (core dumped)
$srcpath/configure "$@" --disable-shared --with-pic
configure: error: ./configure.gnu failed for libs/libedit

Any idea what's wrong here. I did tried to run configure without any
parameters but same result.

Thank you.


-- 
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 lists at kavun.ch  Fri Sep 21 02:26:28 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 20 Sep 2012 18:26:28 -0400
Subject: [Freeswitch-users] bridge_terminate_key alternatives?
Message-ID: <6DC17BCB-4E57-4A5B-9C09-F9F8D32E0C2A@kavun.ch>

Hello listers,

bridge_terminate_key is good for what I am trying to do, except the limitation to only 1 key.
Is there a way so to be able to terminate a bridge session with a DTMF sequence? E.g.: *#.
As a fallback solution, I think I could put in a feature extension that kills the UUID of the leg I need to hangup. Is that the route you would recommend?

Cheers,
Emrah

From msc at freeswitch.org  Fri Sep 21 03:28:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Sep 2012 16:28:25 -0700
Subject: [Freeswitch-users] UK English Prompt Set Recording
In-Reply-To: <CAPerZE9uebr-hGnO-YzKoR-qYxAP9UPVCeO7+4BVJL+4Nbz=jQ@mail.gmail.com>
References: <CAA8_NKD3EqRt6Csy67ADfHx6dDP=19qtA0ob1+_JSo7ifYeMRg@mail.gmail.com>
	<CC7F721F.4D40B%krice@freeswitch.org>
	<CALNZuwmNpzo=z73Y2QF4vgjjOav0cwUjCmk3q501mucqJ8R5-g@mail.gmail.com>
	<CAMbW9o+PBYiX6SV1f0caokiw1wLQPagACB1PBen+4bx3_4hTXA@mail.gmail.com>
	<CAKzWOxU55GCJY7XLh0ECkwk==+Yan87VgYyx-m-O9nm0-ORyrA@mail.gmail.com>
	<CAPerZE9uebr-hGnO-YzKoR-qYxAP9UPVCeO7+4BVJL+4Nbz=jQ@mail.gmail.com>
Message-ID: <CAKzWOxUM7gjm15Q5W+KC3O0=31KA3CTjzh=8u5aO-WowMEz1Gw@mail.gmail.com>

Campbell,

Thank you so much for taking the lead on this. To all those who are even
remotely interested in an en_UK sound set please contact Campbell and
pledge a few bucks/quid/Euros/Krugerrands/etc. :)

-MC

On Thu, Sep 20, 2012 at 2:50 PM, Campbell Steven <casteven at gmail.com> wrote:

> Hi,
>
> So far I have 3 people (from this thread) that have shown interest, if
> I include myself then we have 4, I will respond offlist to those who
> are interested in going further and see what we can come up with.
>
> I have gone down the path of holding a private prompt set before, and
> it's a laborious and difficult project I don't wish to do again, and
> want to help the rest of the community avoid it. With GM Voices
> everyone can get their own custom prompts with matching talent going
> forward.
>
> Campbell
>
> On 21 September 2012 08:56, Michael Collins <msc at freeswitch.org> wrote:
> > Well, only one or two people have shown any interest at all and AFAIK we
> > haven't had any pledge any money, although Campbell can update us on
> that.
> > Keep in mind that you don't need to know the final price if you just
> want to
> > pledge $25 or whatever to the final cost. A lot of little contributions
> will
> > add up, so don't be shy!
> >
> > -MC
> >
> >
> > On Thu, Sep 20, 2012 at 1:34 PM, Andrew Cassidy
> > <andrew at cassidywebservices.co.uk> wrote:
> >>
> >>
> >> Strangely enough I was talking to the wife about trying this the other
> >> day. Not professionally though, haven't got the funding.
> >>
> >> On 19 Sep 2012 19:10, "Avi Marcus" <avi at avimarcus.net> wrote:
> >>>
> >>> Could we get a quote and a pledge list, to see how short we are?
> >>>
> >>> -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
>



-- 
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 Sep 21 03:35:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Sep 2012 16:35:13 -0700
Subject: [Freeswitch-users] bridge_terminate_key alternatives?
In-Reply-To: <6DC17BCB-4E57-4A5B-9C09-F9F8D32E0C2A@kavun.ch>
References: <6DC17BCB-4E57-4A5B-9C09-F9F8D32E0C2A@kavun.ch>
Message-ID: <CAKzWOxXXYiFkU9KTXwZTPNSagFDdk2sXv0Ca8qT92N4AupWpQA@mail.gmail.com>

You could use bind_digit_action, perhaps like this:

<action application="bind_digit_action" data="hangup,*#,exec:hangup,BUH-bye"/>
<action application="digit_action_set_realm" data="hangup"/>

-MC

On Thu, Sep 20, 2012 at 3:26 PM, Emrah <lists at kavun.ch> wrote:

> Hello listers,
>
> bridge_terminate_key is good for what I am trying to do, except the
> limitation to only 1 key.
> Is there a way so to be able to terminate a bridge session with a DTMF
> sequence? E.g.: *#.
> As a fallback solution, I think I could put in a feature extension that
> kills the UUID of the leg I need to hangup. Is that the route you would
> recommend?
>
> Cheers,
> 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 bdfoster at endigotech.com  Fri Sep 21 03:47:30 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 20 Sep 2012 19:47:30 -0400
Subject: [Freeswitch-users] recording's channels shifted
In-Reply-To: <CAKzWOxX2VfzHyAmsTtDQo8MWNWUx3-4=WqNkVMoLSZm-yazu4g@mail.gmail.com>
References: <505B8495.5060601@freemail.hu>
	<CAKzWOxX2VfzHyAmsTtDQo8MWNWUx3-4=WqNkVMoLSZm-yazu4g@mail.gmail.com>
Message-ID: <CAKV-jFyVm-C-_SkZxDZiKd-wiunkLWp8VHVUyzQod_VEgnGR0w@mail.gmail.com>

I've ran into this before, and it ended up being feedback from the handsets
on both sides of the call. It doesn't really bother us now that we know
what it is.

-BDF

On Thu, Sep 20, 2012 at 5:24 PM, Michael Collins <msc at freeswitch.org> wrote:

> I recommend that you get a pcap of such a conversation and analyze it in
> Wireshark. See if the lag shows up in the analysis or not. That will at
> least give you a clue as to where the issue is. (Compare the recording on
> disk with the playback in Wireshark. Note: you'll need the call to be g.711
> for this to work.)
>
> -MC
>
> On Thu, Sep 20, 2012 at 2:03 PM, T?th R?bert <trob at freemail.hu> wrote:
>
>>  No, the callers do not notice anything, they hear each other without problem.
>>
>> One party is on the local network where FS is, and other is outside.
>> But as they do not notice this problem, i think the problem is at the recording. I think.
>>
>>
>> >* *This is more likely an issue with the media streams than the recording
>> >* *itself. Do the callers notice anything? Is one or both callers outside the
>> >* *local network where the FS server reside? Just looking for places where
>> >* *interesting/weird things might happen.
>> >* *
>> >* *-MC
>> >* *
>> >* *On Thu, Sep 20, 2012 at 6:40 AM, T?th R?bert <trob at freemail.hu <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
>> >* *
>> >* *>*  Hi*>* *>**>* *>* Why left and right channels' voice shifted from each other in the recorded*>* *>* mp3?*>* *>**>* *>* I record the sessions' voice thios way:*> >* <action application="set" data="execute_on_answer=record_session*> >* $${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/>*> >* When i listen back the mp3, at begining the two chennels are synchonized,*> >* but after some minutes there will be a shifting between the channels.*> >**> >* Why?*> >**> >* thanks,*> >* Robert*
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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

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From gabe at gundy.org  Fri Sep 21 03:50:33 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Sep 2012 17:50:33 -0600
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: <CAMwZ-tK7p3kZXbnq-WsjzT-ThWZGAk5f9JkUDmVfkSXP4uF2sw@mail.gmail.com>

On Fri, Aug 31, 2012 at 9:33 AM, Michael Collins <msc at freeswitch.org> 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.

I know this is old and most everyone has moved on, but I thought I'd
share this interesting link. I hope that people continue to use the
mailing list in ways that benefit to the community as a whole. I know
I can do better.

http://www.codinghorror.com/blog/2012/03/rubber-duck-problem-solving.html


Best,
Gabe


From mthakershi at gmail.com  Fri Sep 21 04:16:27 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 20 Sep 2012 19:16:27 -0500
Subject: [Freeswitch-users] Speak vs. StreamFile
In-Reply-To: <020f01cd9762$8a60a0b0$9f21e210$@gmail.com>
References: <CAKHK4ZPMn8G=7qByqgBePYVrB-2qXrZBG5ScrBATthQnAzJpgw@mail.gmail.com>
	<CALNZuw=mhmxzBo73cTyaAho8PfwDReBAn6ZDgJwFu7VampJBRw@mail.gmail.com>
	<020f01cd9762$8a60a0b0$9f21e210$@gmail.com>
Message-ID: <CAKHK4ZNpcThYKuDKzJunOags+rM4dnvDE=Xtg1gUL_iDDq0n_w@mail.gmail.com>

Thanks for responses. I will try these options.

But considering design internals of FS,
what do you think is more efficient for playing a static audio file - TTS
or WAV file on disk?

On Thu, Sep 20, 2012 at 2:02 PM, grmt <garmt.noname at gmail.com> wrote:

> If you still want to use TTS, you may want to cache your generated TTS,
> e.g. http://www.tomp.co.uk/blog/view/23****
>
> Note that your TTS license may not allow you to do so ...****
>
> ** **
>
> Alternatively, you can gain some cpu cycles by setting
> cache_speech_handles to true in your dialplan.****
>
> e.g****
>
> <extension name=?use_tts?>****
>
>         <action application="set" data="cache_speech_handles=true"/>****
>
>   ****
>
>                use tts****
>
>                use tts again****
>
> </extension>****
>
> ** **
>
> Not sure how many cycles you will gain though?****
>
> ** **
>
> Garmt****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Thursday, September 20, 2012 20:00
> *To:* FreeSWITCH Users Help
>
> *Subject:* Re: [Freeswitch-users] Speak vs. StreamFile****
>
> ** **
>
> Give it a try... if you've got the volume, even better than wav is native
> file: http://wiki.freeswitch.org/wiki/Mod_native_file
> ****
>
> -Avi****
>
> ** **
>
> On Thu, Sep 20, 2012 at 8:56 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:****
>
> Hello,****
>
> ** **
>
> I am working on optimizing call flow code. I use Cepstral as TTS from FS
> managed module (C# .net code).****
>
> ** **
>
> I suspect using Session.Speak is slowing down the call experience (prompt
> delay) when there are # of concurrent calls.****
>
> ** **
>
> My question: does it help with performance if I create WAV file for each
> static prompt in advance and use Session.StreamFile instead of
> Session.Speak?****
>
> ** **
>
> Any help is appreciated.****
>
> ** **
>
> Malay****
>
> ** **
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 21 04:19:04 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Sep 2012 21:19:04 -0300
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <CAMwZ-tK7p3kZXbnq-WsjzT-ThWZGAk5f9JkUDmVfkSXP4uF2sw@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>
	<CAMwZ-tK7p3kZXbnq-WsjzT-ThWZGAk5f9JkUDmVfkSXP4uF2sw@mail.gmail.com>
Message-ID: <CABH8QX3QA_fzvFmawEWJvm371e-JR+3k5OzNUbcunn74J85H1w@mail.gmail.com>

Duck testing is widely used by all of us on problem solving in general, not
just developers. Nonetheless, they could've found a less fortunate name
don't you think?

:D

Jo?o Mesquita



On Thu, Sep 20, 2012 at 8:50 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Aug 31, 2012 at 9:33 AM, Michael Collins <msc at freeswitch.org>
> 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.
>
> I know this is old and most everyone has moved on, but I thought I'd
> share this interesting link. I hope that people continue to use the
> mailing list in ways that benefit to the community as a whole. I know
> I can do better.
>
> http://www.codinghorror.com/blog/2012/03/rubber-duck-problem-solving.html
>
>
> 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 bdfoster at endigotech.com  Fri Sep 21 06:52:28 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 20 Sep 2012 22:52:28 -0400
Subject: [Freeswitch-users] Segmentation Fault while running configure
In-Reply-To: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>
References: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>
Message-ID: <CAKV-jFwtCtBk+K1nx60_DAPDosTVXMRnyMM7iaYPf0n9gEJ-uQ@mail.gmail.com>

http://jira.freeswitch.org

On Thu, Sep 20, 2012 at 6:04 PM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> Hi,
>
> I am running Ubuntu 12.04 64-bit in KVM and getting segmentation fault
> while running configure script of latest stable Freeswitch v1.2. Here are
> lines just before crash.
>
> /configure --enable-64 --enable-core-odbc-support --enable-optimization
> --with-pic
> ...
> checking whether getpwnam_r and getpwuid_r are posix like... yes
> checking whether getpwnam_r and getpwuid_r are posix _draft_ like... no
> configure: creating ./config.status
> ./configure.gnu: line 3: 20996 Segmentation fault      (core dumped)
> $srcpath/configure "$@" --disable-shared --with-pic
> configure: error: ./configure.gnu failed for libs/libedit
>
> Any idea what's wrong here. I did tried to run configure without any
> parameters but same result.
>
> Thank you.
>
>
> --
> 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
>
>


-- 
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 peter.olsson at visionutveckling.se  Fri Sep 21 07:10:25 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 21 Sep 2012 03:10:25 +0000
Subject: [Freeswitch-users] Segmentation Fault while running configure
In-Reply-To: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>
References: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>
Message-ID: <bebkp1kv6t0sod4waf5wc41h.1348196886298@email.android.com>

Sometimes this happens if you've lowered the stack size in the running shell. Try to logout and login into a new session and see if that works.


/Peter

Muhammad Shahzad <shaheryarkh at googlemail.com> skrev:
Hi,

I am running Ubuntu 12.04 64-bit in KVM and getting segmentation fault while running configure script of latest stable Freeswitch v1.2. Here are lines just before crash.

/configure --enable-64 --enable-core-odbc-support --enable-optimization --with-pic
...
checking whether getpwnam_r and getpwuid_r are posix like... yes
checking whether getpwnam_r and getpwuid_r are posix _draft_ like... no
configure: creating ./config.status
./configure.gnu: line 3: 20996 Segmentation fault      (core dumped) $srcpath/configure "$@" --disable-shared --with-pic
configure: error: ./configure.gnu failed for libs/libedit

Any idea what's wrong here. I did tried to run configure without any parameters but same result.

Thank you.


--
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<mailto:shari_786pk at hotmail.com>
Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
!DSPAM:505b903032762130212658!


From brian at freeswitch.org  Fri Sep 21 07:15:46 2012
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Sep 2012 22:15:46 -0500
Subject: [Freeswitch-users] Segmentation Fault while running configure
In-Reply-To: <CAKV-jFwtCtBk+K1nx60_DAPDosTVXMRnyMM7iaYPf0n9gEJ-uQ@mail.gmail.com>
References: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>
	<CAKV-jFwtCtBk+K1nx60_DAPDosTVXMRnyMM7iaYPf0n9gEJ-uQ@mail.gmail.com>
Message-ID: <4599071577969957374@unknownmsgid>

Enable optimization? Really?

--
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 Sep 20, 2012, at 9:55 PM, Brian Foster <bdfoster at endigotech.com> wrote:

http://jira.freeswitch.org

On Thu, Sep 20, 2012 at 6:04 PM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> Hi,
>
> I am running Ubuntu 12.04 64-bit in KVM and getting segmentation fault
> while running configure script of latest stable Freeswitch v1.2. Here are
> lines just before crash.
>
> /configure --enable-64 --enable-core-odbc-support --enable-optimization
> --with-pic
> ...
> checking whether getpwnam_r and getpwuid_r are posix like... yes
> checking whether getpwnam_r and getpwuid_r are posix _draft_ like... no
> configure: creating ./config.status
> ./configure.gnu: line 3: 20996 Segmentation fault      (core dumped)
> $srcpath/configure "$@" --disable-shared --with-pic
> configure: error: ./configure.gnu failed for libs/libedit
>
> Any idea what's wrong here. I did tried to run configure without any
> parameters but same result.
>
> Thank you.
>
>
> --
> 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
>
>


-- 
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
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From lists at kavun.ch  Fri Sep 21 07:33:28 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 20 Sep 2012 23:33:28 -0400
Subject: [Freeswitch-users] bridge_terminate_key alternatives?
In-Reply-To: <CAKzWOxXXYiFkU9KTXwZTPNSagFDdk2sXv0Ca8qT92N4AupWpQA@mail.gmail.com>
References: <6DC17BCB-4E57-4A5B-9C09-F9F8D32E0C2A@kavun.ch>
	<CAKzWOxXXYiFkU9KTXwZTPNSagFDdk2sXv0Ca8qT92N4AupWpQA@mail.gmail.com>
Message-ID: <1310A84D-457C-4B3D-836F-FFA9FDB4EFCF@kavun.ch>

MC, you are great! I'll be playing with bind_digit_action now.

Thanks for helping. I've learned from every single one of your posts I read on this list.

Active contributors on this list could have a personal donation page for those of us who want to show some gratitude for all the effort you guys are putting in. 


On Sep 20, 2012, at 7:35 PM, Michael Collins <msc at freeswitch.org> wrote:

> You could use bind_digit_action, perhaps like this:
> 
> <action application="bind_digit_action" data="hangup,*#,exec:hangup,BUH-bye"/>
> <action application="digit_action_set_realm" data="hangup"/>
> 
> -MC
> 
> On Thu, Sep 20, 2012 at 3:26 PM, Emrah <lists at kavun.ch> wrote:
> Hello listers,
> 
> bridge_terminate_key is good for what I am trying to do, except the limitation to only 1 key.
> Is there a way so to be able to terminate a bridge session with a DTMF sequence? E.g.: *#.
> As a fallback solution, I think I could put in a feature extension that kills the UUID of the leg I need to hangup. Is that the route you would recommend?
> 
> Cheers,
> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anton.jugatsu at gmail.com  Fri Sep 21 11:57:44 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 21 Sep 2012 11:57:44 +0400
Subject: [Freeswitch-users] Slides from reSIProcate/repro presentation @
	weekly conference talk (19.09)
Message-ID: <CANBhyPCyZ1zUetHUqNqfqhAaxfUCSYVWu8kkjHUY4ZGOzAkRyw@mail.gmail.com>

Guys, I will really appreciate if someone will share this slides/pdf
directly me to email or to file hosting.
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From ben at langfeld.co.uk  Fri Sep 21 12:19:31 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 21 Sep 2012 09:19:31 +0100
Subject: [Freeswitch-users] Speak vs. StreamFile
In-Reply-To: <CAKHK4ZNpcThYKuDKzJunOags+rM4dnvDE=Xtg1gUL_iDDq0n_w@mail.gmail.com>
References: <CAKHK4ZPMn8G=7qByqgBePYVrB-2qXrZBG5ScrBATthQnAzJpgw@mail.gmail.com>
	<CALNZuw=mhmxzBo73cTyaAho8PfwDReBAn6ZDgJwFu7VampJBRw@mail.gmail.com>
	<020f01cd9762$8a60a0b0$9f21e210$@gmail.com>
	<CAKHK4ZNpcThYKuDKzJunOags+rM4dnvDE=Xtg1gUL_iDDq0n_w@mail.gmail.com>
Message-ID: <CAAyX+KHQaQg5g_vYynd31L6XqG1c4_jP9u2EjFer=6zef-hCag@mail.gmail.com>

Well, remembering that TTS requires per-channel licensing, I would say
avoid using it where possible to increase your call density per TTS channel
license.

Regards,
Ben Langfeld


On 21 September 2012 01:16, Malay Thakershi <mthakershi at gmail.com> wrote:

> Thanks for responses. I will try these options.
>
> But considering design internals of FS,
> what do you think is more efficient for playing a static audio file - TTS
> or WAV file on disk?
>
> On Thu, Sep 20, 2012 at 2:02 PM, grmt <garmt.noname at gmail.com> wrote:
>
>> If you still want to use TTS, you may want to cache your generated TTS,
>> e.g. http://www.tomp.co.uk/blog/view/23****
>>
>> Note that your TTS license may not allow you to do so ...****
>>
>> ** **
>>
>> Alternatively, you can gain some cpu cycles by setting
>> cache_speech_handles to true in your dialplan.****
>>
>> e.g****
>>
>> <extension name=?use_tts?>****
>>
>>         <action application="set" data="cache_speech_handles=true"/>****
>>
>>   ****
>>
>>                use tts****
>>
>>                use tts again****
>>
>> </extension>****
>>
>> ** **
>>
>> Not sure how many cycles you will gain though?****
>>
>> ** **
>>
>> Garmt****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
>> *Sent:* Thursday, September 20, 2012 20:00
>> *To:* FreeSWITCH Users Help
>>
>> *Subject:* Re: [Freeswitch-users] Speak vs. StreamFile****
>>
>> ** **
>>
>> Give it a try... if you've got the volume, even better than wav is native
>> file: http://wiki.freeswitch.org/wiki/Mod_native_file
>> ****
>>
>> -Avi****
>>
>> ** **
>>
>> On Thu, Sep 20, 2012 at 8:56 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:****
>>
>> Hello,****
>>
>> ** **
>>
>> I am working on optimizing call flow code. I use Cepstral as TTS from FS
>> managed module (C# .net code).****
>>
>> ** **
>>
>> I suspect using Session.Speak is slowing down the call experience (prompt
>> delay) when there are # of concurrent calls.****
>>
>> ** **
>>
>> My question: does it help with performance if I create WAV file for each
>> static prompt in advance and use Session.StreamFile instead of
>> Session.Speak?****
>>
>> ** **
>>
>> Any help is appreciated.****
>>
>> ** **
>>
>> Malay****
>>
>> ** **
>>
>> ** **
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shaheryarkh at googlemail.com  Fri Sep 21 12:51:30 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 21 Sep 2012 10:51:30 +0200
Subject: [Freeswitch-users] Segmentation Fault while running configure
In-Reply-To: <bebkp1kv6t0sod4waf5wc41h.1348196886298@email.android.com>
References: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>
	<bebkp1kv6t0sod4waf5wc41h.1348196886298@email.android.com>
Message-ID: <CAFZQphxoBp=bt3TTYb=yBvfk75NHBjbjVSAgb-GbC=H4XZ7KdQ@mail.gmail.com>

This makes a lot of sense. I will check and get back to you.

Thank you.

On Sep 21, 2012 5:11 AM, "Peter Olsson" <peter.olsson at visionutveckling.se>
wrote:
>
> Sometimes this hap
>
> Sometimes this happens if you've lowered the stack size in the running
> shell. Try to logout and login into a new session and see if that works.
>
>
> /Peter
>
> Muhammad Shahzad <shaheryarkh at googlemail.com> skrev:
> Hi,
>
> I am running Ubuntu 12.04 64-bit in KVM and getting segmentation fault
> while running configure script of latest stable Freeswitch v1.2. Here are
> lines just before crash.
>
> /configure --enable-64 --enable-core-odbc-support --enable-optimization
> --with-pic
> ...
> checking whether getpwnam_r and getpwuid_r are posix like... yes
> checking whether getpwnam_r and getpwuid_r are posix _draft_ like... no
> configure: creating ./config.status
> ./configure.gnu: line 3: 20996 Segmentation fault      (core dumped)
> $srcpath/configure "$@" --disable-shared --with-pic
> configure: error: ./configure.gnu failed for libs/libedit
>
> Any idea what's wrong here. I did tried to run configure without any
> parameters but same result.
>
> Thank you.
>
>
> --
> 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<mailto:shari_786pk at hotmail.com>
> Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
> !DSPAM:505b903032762130212658!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shaheryarkh at googlemail.com  Fri Sep 21 12:54:40 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 21 Sep 2012 10:54:40 +0200
Subject: [Freeswitch-users] Segmentation Fault while running configure
In-Reply-To: <4599071577969957374@unknownmsgid>
References: <CAFZQphxBFTzJHjORLTX1AGkk+3sDyLJLQLgdEj594M3pPGZQ6w@mail.gmail.com>
	<CAKV-jFwtCtBk+K1nx60_DAPDosTVXMRnyMM7iaYPf0n9gEJ-uQ@mail.gmail.com>
	<4599071577969957374@unknownmsgid>
Message-ID: <CAFZQphyU4sQsKh2XmK27JCt5_jTbDP3Rh0+gc4Dqu=fh1T7ZXw@mail.gmail.com>

What is wrong with enabling optimation?

On Sep 21, 2012 5:16 AM, "Brian West" <brian at freeswitch.org> wrote:
>
> Enable optimization?
>
> Enable optimization? Really?
>
> --
> 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 Sep 20, 2012, at 9:55 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>
> http://jira.freeswitch.org
>
> On Thu, Sep 20, 2012 at 6:04 PM, Muhammad Shahzad <
> shaheryarkh at googlemail.com> wrote:
>
>> Hi,
>>
>> I am running Ubuntu 12.04 64-bit in KVM and getting segmentation fault
>> while running configure script of latest stable Freeswitch v1.2. Here are
>> lines just before crash.
>>
>> /configure --enable-64 --enable-core-odbc-support --enable-optimization
>> --with-pic
>> ...
>> checking whether getpwnam_r and getpwuid_r are posix like... yes
>> checking whether getpwnam_r and getpwuid_r are posix _draft_ like... no
>> configure: creating ./config.status
>> ./configure.gnu: line 3: 20996 Segmentation fault      (core dumped)
>> $srcpath/configure "$@" --disable-shared --with-pic
>> configure: error: ./configure.gnu failed for libs/libedit
>>
>> Any idea what's wrong here. I did tried to run configure without any
>> parameters but same result.
>>
>> Thank you.
>>
>>
>> --
>> 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
>>
>>
>
>
> --
> 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
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From trob at freemail.hu  Fri Sep 21 13:14:32 2012
From: trob at freemail.hu (=?ISO-8859-1?Q?T=F3th_R=F3bert?=)
Date: Fri, 21 Sep 2012 11:14:32 +0200
Subject: [Freeswitch-users] recording's channels shifted
In-Reply-To: <CAKV-jFyVm-C-_SkZxDZiKd-wiunkLWp8VHVUyzQod_VEgnGR0w@mail.gmail.com>
References: <505B8495.5060601@freemail.hu>
	<CAKzWOxX2VfzHyAmsTtDQo8MWNWUx3-4=WqNkVMoLSZm-yazu4g@mail.gmail.com>
	<CAKV-jFyVm-C-_SkZxDZiKd-wiunkLWp8VHVUyzQod_VEgnGR0w@mail.gmail.com>
Message-ID: <505C2FF8.1030409@freemail.hu>

Could you explain me this?



Felad?: Brian Foster <bdfoster at endigotech.com>
C?mzett: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
T?rgy: Re: [Freeswitch-users] recording's channels shifted
D?tum: 2012. Szeptember 21. 1:47:30
> I've ran into this before, and it ended up being feedback from the handsets on both sides of the 
> call. It doesn't really bother us now that we know what it is.
>
> -BDF
>
> On Thu, Sep 20, 2012 at 5:24 PM, Michael Collins <msc at freeswitch.org <mailto:msc at freeswitch.org>> 
> wrote:
>
>     I recommend that you get a pcap of such a conversation and analyze it in Wireshark. See if the
>     lag shows up in the analysis or not. That will at least give you a clue as to where the issue
>     is. (Compare the recording on disk with the playback in Wireshark. Note: you'll need the call
>     to be g.711 for this to work.)
>
>     -MC
>
>     On Thu, Sep 20, 2012 at 2:03 PM, T?th R?bert <trob at freemail.hu <mailto:trob at freemail.hu>> wrote:
>
>         No, the callers do not notice anything, they hear each other without problem.
>
>         One party is on the local network where FS is, and other is outside.
>         But as they do not notice this problem, i think the problem is at the recording. I think.
>
>
>         >/  /This is more likely an issue with the media streams than the recording
>         >/  /itself. Do the callers notice anything? Is one or both callers outside the
>         >/  /local network where the FS server reside? Just looking for places where
>         >/  /interesting/weird things might happen.
>         >/  /
>         >/  /-MC
>         >/  /
>         >/  /On Thu, Sep 20, 2012 at 6:40 AM, T?th R?bert <trob at freemail.hu  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
>         >/  /
>         >/  />/   Hi
>         />/  />/
>         />/  />/  Why left and right channels' voice shifted from each other in the recorded
>         />/  />/  mp3?
>         />/  />/
>         />/  />/  I record the sessions' voice thios way:
>         /> >/  <action application="set" data="execute_on_answer=record_session
>         /> >/  $${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/>
>         /> >/  When i listen back the mp3, at begining the two chennels are synchonized,
>         /> >/  but after some minutes there will be a shifting between the channels.
>         /> >/
>         /> >/  Why?
>         /> >/
>         /> >/  thanks,
>         /> >/  Robert/
>
>
>         _________________________________________________________________________
>         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
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>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     -- 
>     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
>
>
>
>
> -- 
> 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
> 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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From vipkilla at gmail.com  Fri Sep 21 17:12:11 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 21 Sep 2012 09:12:11 -0400
Subject: [Freeswitch-users] sofia recover does not recover calls on hold
Message-ID: <CAC-LwPNCXzB_rmM3pon5wkU8=sAe84hMNjJYRQOKykU=i9T3ew@mail.gmail.com>

Hi,
I've been testing 'sofia recover' the past few days using two
freeswitch servers and a floating IP.
It works great in general, but I did notice a problem with any call
that is holding during the crash and recovery.
Basically, any call that is on hold comes back after 'sofia recover'
but when trying to take the call off hold the call is dropped.
I'm using the following parameters on both FS instances in switch.conf.xml:

    <param name="switchname" value="fs01"/>
    <param name="core-db-dsn" value="dsn:name:password"/>
    <param name="core-recovery-db-dsn" value="dsn:name:password"/>

And in sofia config:

<param name="track-calls" value="true"/>
<param name="odbc-dsn" value="dsn:name:password"/>

As I said before, sofia recovery works except for the calls
holding...just wondering if this is a known issue before I open up a
jira ticket.
Thanks!
-Vik


From vitaliy.davudov at vts24.ru  Fri Sep 21 17:42:05 2012
From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=)
Date: Fri, 21 Sep 2012 17:42:05 +0400
Subject: [Freeswitch-users] Git checkout failed
Message-ID: <505C6EAD.5070204@vts24.ru>

Hi, list!

I try to update my freeswitch to stable version 1.2. According to FS 
Wiki 
(http://wiki.freeswitch.org/wiki/Installation_Guide#Selecting_version) I 
applied this command in previos crated freeswitch.git directory:

git checkout v1.2.stable

But system returned me this error:

error: pathspec 'v1.2.stable' did not match any file(s) known to git.

What did I do wrong?

-- 
Best regards,
Vitaly.



From avi at avimarcus.net  Fri Sep 21 17:57:15 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 21 Sep 2012 16:57:15 +0300
Subject: [Freeswitch-users] Git checkout failed
In-Reply-To: <505C6EAD.5070204@vts24.ru>
References: <505C6EAD.5070204@vts24.ru>
Message-ID: <CALNZuw=ezZXnXBGn43ogXeMotKD55i17_4kM9qtP3zB5RRGykA@mail.gmail.com>

Try a git pull first.
-Avi



On Fri, Sep 21, 2012 at 4:42 PM, ??????? ???????
<vitaliy.davudov at vts24.ru>wrote:

> Hi, list!
>
> I try to update my freeswitch to stable version 1.2. According to FS
> Wiki
> (http://wiki.freeswitch.org/wiki/Installation_Guide#Selecting_version) I
> applied this command in previos crated freeswitch.git directory:
>
> git checkout v1.2.stable
>
> But system returned me this error:
>
> error: pathspec 'v1.2.stable' did not match any file(s) known to git.
>
> What did I do wrong?
>
> --
> Best regards,
> Vitaly.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Fri Sep 21 17:58:55 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 21 Sep 2012 08:58:55 -0500
Subject: [Freeswitch-users] Git checkout failed
In-Reply-To: <505C6EAD.5070204@vts24.ru>
Message-ID: <CC81DCCF.5317A%krice@freeswitch.org>

Sounds like you might have an older version of git... Try a git fetch --all
then a git checkout v1.2.stable


On 9/21/12 8:42 AM, "??????? ???????" <vitaliy.davudov at vts24.ru> wrote:

> Hi, list!
> 
> I try to update my freeswitch to stable version 1.2. According to FS
> Wiki 
> (http://wiki.freeswitch.org/wiki/Installation_Guide#Selecting_version) I
> applied this command in previos crated freeswitch.git directory:
> 
> git checkout v1.2.stable
> 
> But system returned me this error:
> 
> error: pathspec 'v1.2.stable' did not match any file(s) known to git.
> 
> What did I do wrong?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From ben at langfeld.co.uk  Fri Sep 21 18:00:14 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 21 Sep 2012 15:00:14 +0100
Subject: [Freeswitch-users] Git checkout failed
In-Reply-To: <505C6EAD.5070204@vts24.ru>
References: <505C6EAD.5070204@vts24.ru>
Message-ID: <CAAyX+KEpLihKCJvRbhqr5d81E-NKcY4Qfai68_99zdCsHD6EMw@mail.gmail.com>

Try:

git fetch origin
git checkout origin/v1.2.stable

Regards,
Ben Langfeld


On 21 September 2012 14:42, ??????? ??????? <vitaliy.davudov at vts24.ru>wrote:

> Hi, list!
>
> I try to update my freeswitch to stable version 1.2. According to FS
> Wiki
> (http://wiki.freeswitch.org/wiki/Installation_Guide#Selecting_version) I
> applied this command in previos crated freeswitch.git directory:
>
> git checkout v1.2.stable
>
> But system returned me this error:
>
> error: pathspec 'v1.2.stable' did not match any file(s) known to git.
>
> What did I do wrong?
>
> --
> Best regards,
> Vitaly.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From b2m at a-cti.com  Fri Sep 21 18:10:17 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Fri, 21 Sep 2012 19:40:17 +0530
Subject: [Freeswitch-users] FreeSWITCH : mod_skypopen
Message-ID: <CAP2fbbda3qpPOhJ+kUNU6B-F3MxwDZ6ebeBW3w1ZnPwBD4yngQ@mail.gmail.com>

Is it possible to handle Skype chat and presence through lua script?

Thanks,
Bala
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From gmaruzz at gmail.com  Fri Sep 21 18:24:50 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 21 Sep 2012 16:24:50 +0200
Subject: [Freeswitch-users] FreeSWITCH : mod_skypopen
In-Reply-To: <CAP2fbbda3qpPOhJ+kUNU6B-F3MxwDZ6ebeBW3w1ZnPwBD4yngQ@mail.gmail.com>
References: <CAP2fbbda3qpPOhJ+kUNU6B-F3MxwDZ6ebeBW3w1ZnPwBD4yngQ@mail.gmail.com>
Message-ID: <CALXCt0rHDtcv=4YQTqd5xpQS20oei=H6Bdn6=CP-TtUMHgBNHg@mail.gmail.com>

You can use the chatplan for chat (load mod_sms for this) and from
chatplan you can use lua.

Presence is not managed at the moment.

If you are thinking at the presence of "others" (eg: the buddy list)
this is not managed at the moment.

If you are thinking at the presence of himself (eg: changing the
presence of the interface from online to busy or away or do not
disturb) it's not directly managed (it's forced to "online" at
startup), but you can do that by sending commands to the skype client:
http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#skypopen
"SET USERSTATUS ONLINE"

from http://developer.skype.com/public-api-reference :
GET USERSTATUS

This command queries or modifies user visiblity for the current user.

Syntax
GET USERSTATUS
SET USERSTATUS <value>

Response
USERSTATUS <value>

Parameters
<value> ? new userstatus. Possible values:

UNKNOWN
ONLINE ? current user is online
OFFLINE ? current user is offline
SKYPEME ? current user is in ?Skype Me? mode (protocol 2).
AWAY ? current user is away.
NA ? current user is not available.
DND ? current user is in ?Do not disturb? mode.
INVISIBLE ? current user is invisible to others.
LOGGEDOUT ? current user is logged out. Clients are detached.

Version
Protocol 1

Errors

ERROR 28 Unknown userstatus
Status value is incorrect or misspelled

Example

-> SET USERSTATUS OFFLINE
<- USERSTATUS OFFLINE
<- USERSTATUS OFFLINE
-> SET USERSTATUS xxx
<- ERROR 28 Unknown userstatus



On Fri, Sep 21, 2012 at 4:10 PM, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>
> Is it possible to handle Skype chat and presence through lua script?
>
> Thanks,
> Bala
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Fri Sep 21 20:09:02 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Sep 2012 11:09:02 -0500
Subject: [Freeswitch-users] sofia recover does not recover calls on hold
In-Reply-To: <CAC-LwPNCXzB_rmM3pon5wkU8=sAe84hMNjJYRQOKykU=i9T3ew@mail.gmail.com>
References: <CAC-LwPNCXzB_rmM3pon5wkU8=sAe84hMNjJYRQOKykU=i9T3ew@mail.gmail.com>
Message-ID: <CAKbxfG9DN_Gb=+n2nk93oSsz3AKmDoLzLSLFJAqsKSp5-xFs7g@mail.gmail.com>

This is a known limitation for the time being.  You can open a JIRA as a
placeholder but it will be a while and some work on the sofia sip stack to
get it working.


On Fri, Sep 21, 2012 at 8:12 AM, Vik Killa <vipkilla at gmail.com> wrote:

> Hi,
> I've been testing 'sofia recover' the past few days using two
> freeswitch servers and a floating IP.
> It works great in general, but I did notice a problem with any call
> that is holding during the crash and recovery.
> Basically, any call that is on hold comes back after 'sofia recover'
> but when trying to take the call off hold the call is dropped.
> I'm using the following parameters on both FS instances in switch.conf.xml:
>
>     <param name="switchname" value="fs01"/>
>     <param name="core-db-dsn" value="dsn:name:password"/>
>     <param name="core-recovery-db-dsn" value="dsn:name:password"/>
>
> And in sofia config:
>
> <param name="track-calls" value="true"/>
> <param name="odbc-dsn" value="dsn:name:password"/>
>
> As I said before, sofia recovery works except for the calls
> holding...just wondering if this is a known issue before I open up a
> jira ticket.
> Thanks!
> -Vik
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rsaavedra at ecogizmos.com  Fri Sep 21 20:14:12 2012
From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com)
Date: Fri, 21 Sep 2012 11:14:12 -0500
Subject: [Freeswitch-users] Change the DID
Message-ID: <fed4e1549c758c9ef7b3069b07d4cb2d.squirrel@emailmg.globat.com>

Hello,

I have a problem with the DID of a SIP trunk between our central
office(using Trixbox) and the new branch in the same city (using
FreeSWITCH).

When a call is made from FreeSWITCH to the Trixbox, the Trixbox receives
$1 in the DID (FROM-DID variable).

How can I specified a different value for the DID?

Thank you,

Ricardo Saavedra




From curriegrad2004 at gmail.com  Fri Sep 21 20:36:55 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 21 Sep 2012 09:36:55 -0700
Subject: [Freeswitch-users] Change the DID
In-Reply-To: <fed4e1549c758c9ef7b3069b07d4cb2d.squirrel@emailmg.globat.com>
References: <fed4e1549c758c9ef7b3069b07d4cb2d.squirrel@emailmg.globat.com>
Message-ID: <CAMWiMy_uA4yJeShLDX=S5bV6dL+98qLw+jvMrDJ7s2bQqJa5zQ@mail.gmail.com>

What Caller ID are you sending with that particular DID?

On Fri, Sep 21, 2012 at 9:14 AM,  <rsaavedra at ecogizmos.com> wrote:
> Hello,
>
> I have a problem with the DID of a SIP trunk between our central
> office(using Trixbox) and the new branch in the same city (using
> FreeSWITCH).
>
> When a call is made from FreeSWITCH to the Trixbox, the Trixbox receives
> $1 in the DID (FROM-DID variable).
>
> How can I specified a different value for the DID?
>
> Thank you,
>
> Ricardo Saavedra
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Sep 21 21:26:25 2012
From: ajohnston at blimessaging.com (Adam Johnston)
Date: Fri, 21 Sep 2012 13:26:25 -0400
Subject: [Freeswitch-users] using spandsp fax tone detection from originate
	command
Message-ID: <CAE2A9TDoXMEhALC5dwBd+1m8X-D2pr278nz-opgLvwVu2MBWZA@mail.gmail.com>

Hi all,

I've been trying to use the spandsp_start_fax_detect application (with
txfax as the callback) from the originate command but I keep getting a
hangup before it completes.

So far I've tried calling the app directly:
originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234
&spandsp_start_fax_detect('txfax /tmp/myfax.tif 3 ced')

Inline dialplan:
originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234
'spandsp_start_fax_detect:txfax /tmp/myfax.tif 3 ced' inline

..plus some other permutations but all end with the same result:

[NOTICE] mod_spandsp.c:107 Enabling fax detection 'txfax' '/tmp/myfax.tif'
[DEBUG] switch_core_media_bug.c:456 Attaching BUG to
sofia/external/18005551234
[NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has
executed the last dialplan instruction, hanging up.

Am I misunderstanding how spandsp_start_fax_detect works?

Adam Johnston
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From msc at freeswitch.org  Fri Sep 21 21:41:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Sep 2012 10:41:59 -0700
Subject: [Freeswitch-users] bridge_terminate_key alternatives?
In-Reply-To: <1310A84D-457C-4B3D-836F-FFA9FDB4EFCF@kavun.ch>
References: <6DC17BCB-4E57-4A5B-9C09-F9F8D32E0C2A@kavun.ch>
	<CAKzWOxXXYiFkU9KTXwZTPNSagFDdk2sXv0Ca8qT92N4AupWpQA@mail.gmail.com>
	<1310A84D-457C-4B3D-836F-FFA9FDB4EFCF@kavun.ch>
Message-ID: <CAKzWOxW0ACNOpGpNt+_jyYm9iFvO65XsV6DD+pWtiN6H46SRqA@mail.gmail.com>

On Thu, Sep 20, 2012 at 8:33 PM, Emrah <lists at kavun.ch> wrote:

> MC, you are great! I'll be playing with bind_digit_action now.
>
> Thanks for helping. I've learned from every single one of your posts I
> read on this list.
>
> Active contributors on this list could have a personal donation page for
> those of us who want to show some gratitude for all the effort you guys are
> putting in.
>

Well, here's a start... :)
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_So_where_do_I_find_wishlists_for_the_developers_of_this_awesome_software.3F

-MC

-- 
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 Sep 21 21:50:48 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Sep 2012 10:50:48 -0700
Subject: [Freeswitch-users] Slides from reSIProcate/repro presentation @
 weekly conference talk (19.09)
In-Reply-To: <CANBhyPCyZ1zUetHUqNqfqhAaxfUCSYVWu8kkjHUY4ZGOzAkRyw@mail.gmail.com>
References: <CANBhyPCyZ1zUetHUqNqfqhAaxfUCSYVWu8kkjHUY4ZGOzAkRyw@mail.gmail.com>
Message-ID: <CAKzWOxW-LGY2pFYuh7W3kWAAWChZ0Agvf_knx1Ez=OS1gdFFoQ@mail.gmail.com>

On Fri, Sep 21, 2012 at 12:57 AM, Anton Kvashenkin
<anton.jugatsu at gmail.com>wrote:

> Guys, I will really appreciate if someone will share this slides/pdf
> directly me to email or to file hosting.
>
No problemo!
http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19#Featured_Presentation

see the "slides" link.
-MC
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From msc at freeswitch.org  Fri Sep 21 22:07:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Sep 2012 11:07:03 -0700
Subject: [Freeswitch-users] using spandsp fax tone detection from
 originate command
In-Reply-To: <CAE2A9TDoXMEhALC5dwBd+1m8X-D2pr278nz-opgLvwVu2MBWZA@mail.gmail.com>
References: <CAE2A9TDoXMEhALC5dwBd+1m8X-D2pr278nz-opgLvwVu2MBWZA@mail.gmail.com>
Message-ID: <CAKzWOxWGCXkKPSvznMxmkEARxYvGSLWf2GQJ8Beds+C7+W+z7A@mail.gmail.com>

Hi Adam,

The key piece of information is this:
[NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has
executed the last dialplan instruction, hanging up.

You've enabled "fax detection" on the call but you haven't told the call to
actually do anything. From what you're doing it seems like you're wanting
to detect a fax and then run the txfax application. But what happens if a
fax is not detected? Also, what happens to the call in the interim, i.e.
between the time you enable fax detect and the time it actually detects a
fax tone? The call needs to do "something" while waiting for a fax tone.
You may be better off sending this call to an extension and then having
several dialplan actions, like "answer", "spandsp_start_fax_detect", and
then a "sleep" or "playback" app, or something for the channel to do while
it's waiting for fax tone.

Question: what do you want the call to do if no fax is detected?

-MC

On Fri, Sep 21, 2012 at 10:26 AM, Adam Johnston
<ajohnston at blimessaging.com>wrote:

> Hi all,
>
> I've been trying to use the spandsp_start_fax_detect application (with
> txfax as the callback) from the originate command but I keep getting a
> hangup before it completes.
>
> So far I've tried calling the app directly:
> originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234&spandsp_start_fax_detect('txfax /tmp/myfax.tif 3 ced')
>
> Inline dialplan:
> originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234'spandsp_start_fax_detect:txfax /tmp/myfax.tif 3 ced' inline
>
> ..plus some other permutations but all end with the same result:
>
> [NOTICE] mod_spandsp.c:107 Enabling fax detection 'txfax' '/tmp/myfax.tif'
> [DEBUG] switch_core_media_bug.c:456 Attaching BUG to sofia/external/
> 18005551234
> [NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has
> executed the last dialplan instruction, hanging up.
>
> Am I misunderstanding how spandsp_start_fax_detect works?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 21 22:21:43 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 21 Sep 2012 13:21:43 -0500
Subject: [Freeswitch-users] Busy week for the FreeSWITCH Crew...
Message-ID: <CC821A67.5319C%krice@freeswitch.org>

Hey Guys,

The FreeSWITCH Crew has had a very busy week.

Things from this week. Security issue in libsofia fixed. The problem was a
specifically crafted Route header on an invite could cause a segfault
leading to a denial of service situation. For those of you that have your
FreeSWITCH boxes exposed to the internet, its a good idea to update at this
time.

This has been resolved in both the master development branch and the stable
branch

Along with resolving this issue and updating the stable branch, we have
released FreeSWITCH 1.2.3.
Source is available in all the normal locations.

Have a great weekend guys!


-- 
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  Fri Sep 21 22:28:41 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 21 Sep 2012 13:28:41 -0500
Subject: [Freeswitch-users] Join us on the FreeSWITCH Conference bridge this
 afternoon for a Community Free For All...
Message-ID: <CC821C09.5319F%krice@freeswitch.org>

Back in the day, sip:888 at conference.freeswitch.org was more than the
location for the Weekly FreeSWITCH conference call. So lets see what we can
get going this afternoon.

Starting at 4PM Eastern Time (that?s noon Pacific or 8PM GMT) join us for a
general call... 

This is no topic and its open for all to join. Use the normal access means
to call on into the conference.

Join us and who knows what the topics will end up being.

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From ajohnston at blimessaging.com  Fri Sep 21 23:31:21 2012
From: ajohnston at blimessaging.com (Adam Johnston)
Date: Fri, 21 Sep 2012 15:31:21 -0400
Subject: [Freeswitch-users] using spandsp fax tone detection from
 originate command
In-Reply-To: <CAKzWOxWGCXkKPSvznMxmkEARxYvGSLWf2GQJ8Beds+C7+W+z7A@mail.gmail.com>
References: <CAE2A9TDoXMEhALC5dwBd+1m8X-D2pr278nz-opgLvwVu2MBWZA@mail.gmail.com>
	<CAKzWOxWGCXkKPSvznMxmkEARxYvGSLWf2GQJ8Beds+C7+W+z7A@mail.gmail.com>
Message-ID: <CAE2A9TBRKxwXOKHH3WwadCt+DUJ0M-sqR5yVWnK2ziP-DitxQg@mail.gmail.com>

I had actually added some sleep commands in some of my other command
combinations but I kept getting the same result. However, once I increased
the timeout to spandsp_start_fax_detect and increased the sleep time I
started getting some successful faxes out.

If no fax tone is detected I would like to set fax_result_code to a custom
error code that I could then map in my controller application. Is there
some way to evaluate the result of a tone detect? I've been trying the
below inline dialplan but it's giving my false negatives so I'm back on the
wiki looking for a solution.

'spandsp_start_fax_detect:txfax /tmp/myfax.tif 10
ced,sleep:10000,set:fax_result_code:951' inline

Many thanks,

Adam Johnston


On Fri, Sep 21, 2012 at 2:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hi Adam,
>
> The key piece of information is this:
>
> [NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has
> executed the last dialplan instruction, hanging up.
>
> You've enabled "fax detection" on the call but you haven't told the call
> to actually do anything. From what you're doing it seems like you're
> wanting to detect a fax and then run the txfax application. But what
> happens if a fax is not detected? Also, what happens to the call in the
> interim, i.e. between the time you enable fax detect and the time it
> actually detects a fax tone? The call needs to do "something" while waiting
> for a fax tone. You may be better off sending this call to an extension and
> then having several dialplan actions, like "answer",
> "spandsp_start_fax_detect", and then a "sleep" or "playback" app, or
> something for the channel to do while it's waiting for fax tone.
>
> Question: what do you want the call to do if no fax is detected?
>
> -MC
>
> On Fri, Sep 21, 2012 at 10:26 AM, Adam Johnston <
> ajohnston at blimessaging.com> wrote:
>
>> Hi all,
>>
>> I've been trying to use the spandsp_start_fax_detect application (with
>> txfax as the callback) from the originate command but I keep getting a
>> hangup before it completes.
>>
>> So far I've tried calling the app directly:
>> originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234&spandsp_start_fax_detect('txfax /tmp/myfax.tif 3 ced')
>>
>> Inline dialplan:
>> originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234'spandsp_start_fax_detect:txfax /tmp/myfax.tif 3 ced' inline
>>
>> ..plus some other permutations but all end with the same result:
>>
>> [NOTICE] mod_spandsp.c:107 Enabling fax detection 'txfax' '/tmp/myfax.tif'
>> [DEBUG] switch_core_media_bug.c:456 Attaching BUG to sofia/external/
>> 18005551234
>> [NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has
>> executed the last dialplan instruction, hanging up.
>>
>> Am I misunderstanding how spandsp_start_fax_detect works?
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 anton.jugatsu at gmail.com  Fri Sep 21 23:41:02 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 21 Sep 2012 23:41:02 +0400
Subject: [Freeswitch-users] Slides from reSIProcate/repro presentation @
 weekly conference talk (19.09)
In-Reply-To: <CAKzWOxW-LGY2pFYuh7W3kWAAWChZ0Agvf_knx1Ez=OS1gdFFoQ@mail.gmail.com>
References: <CANBhyPCyZ1zUetHUqNqfqhAaxfUCSYVWu8kkjHUY4ZGOzAkRyw@mail.gmail.com>
	<CAKzWOxW-LGY2pFYuh7W3kWAAWChZ0Agvf_knx1Ez=OS1gdFFoQ@mail.gmail.com>
Message-ID: <CANBhyPCZHtCDL3xJnaXZDDZ=udfNK38qmGE9pS3XARaFq9Jz4w@mail.gmail.com>

Thanks a lot :)

2012/9/21 Michael Collins <msc at freeswitch.org>

>
>
> On Fri, Sep 21, 2012 at 12:57 AM, Anton Kvashenkin <
> anton.jugatsu at gmail.com> wrote:
>
>> Guys, I will really appreciate if someone will share this slides/pdf
>> directly me to email or to file hosting.
>>
> No problemo!
> http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19#Featured_Presentation
>
> see the "slides" link.
> -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 lists at kavun.ch  Fri Sep 21 23:58:20 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 21 Sep 2012 15:58:20 -0400
Subject: [Freeswitch-users] Join us on the FreeSWITCH Conference bridge
	this afternoon for a Community Free For All...
In-Reply-To: <CC821C09.5319F%krice@freeswitch.org>
References: <CC821C09.5319F%krice@freeswitch.org>
Message-ID: <9EB13CB8-05DA-47D7-8463-5AF8430BA09E@kavun.ch>

I love the idea, but you might have gotten confused with the timezones.
4 PM EDT = 8 PM UTC = 1 PM PDT. 
Is this the timing?

Just dialed in? It's all silence? Can't stay long though.
On Sep 21, 2012, at 2:28 PM, Ken Rice <krice at freeswitch.org> wrote:

> Back in the day, sip:888 at conference.freeswitch.org was more than the location for the Weekly FreeSWITCH conference call. So lets see what we can get going this afternoon.
> 
> Starting at 4PM Eastern Time (that?s noon Pacific or 8PM GMT) join us for a general call... 
> 
> This is no topic and its open for all to join. Use the normal access means to call on into the conference.
> 
> Join us and who knows what the topics will end up being.
> 
> -- 
> 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



From krice at freeswitch.org  Sat Sep 22 00:02:52 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 21 Sep 2012 15:02:52 -0500
Subject: [Freeswitch-users] Join us on the FreeSWITCH Conference bridge
 this afternoon for a Community Free For All...
In-Reply-To: <9EB13CB8-05DA-47D7-8463-5AF8430BA09E@kavun.ch>
Message-ID: <CC82321C.531A9%krice@freeswitch.org>

Oh yeah... I got my PDT calc off.. Its 1PM PDT... If you are having issues
getting unmuted pop on to IRC one of us with mod powers on the conf bridge
will unmute you...

For those using the web client on conference.freeswitch.org please use a
headset to keep the echo down else we'll have to keep you muted to avoid the
problems the echo causes

Thanks
K


On 9/21/12 2:58 PM, "Emrah" <lists at kavun.ch> wrote:

> I love the idea, but you might have gotten confused with the timezones.
> 4 PM EDT = 8 PM UTC = 1 PM PDT.
> Is this the timing?
> 
> Just dialed in? It's all silence? Can't stay long though.
> On Sep 21, 2012, at 2:28 PM, Ken Rice <krice at freeswitch.org> wrote:
> 
>> Back in the day, sip:888 at conference.freeswitch.org was more than the location
>> for the Weekly FreeSWITCH conference call. So lets see what we can get going
>> this afternoon.
>> 
>> Starting at 4PM Eastern Time (that?s noon Pacific or 8PM GMT) join us for a
>> general call... 
>> 
>> This is no topic and its open for all to join. Use the normal access means to
>> call on into the conference.
>> 
>> Join us and who knows what the topics will end up being.
>> 
>> -- 
>> 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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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




From gabe at gundy.org  Sat Sep 22 01:09:20 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 21 Sep 2012 15:09:20 -0600
Subject: [Freeswitch-users] Building v1.2 .debs on Ubuntu 12.04 LTS
Message-ID: <CAMwZ-t+bqZ6NUc5f7z2hFB3dFcNpT=86v3kfH_62Ws=M_Rs6yg@mail.gmail.com>

All,

Before I get started down this path again, I'd like to know if anyone
has a clean list of commands to build FreeSWITCH 1.2 debs on a clean
install of Ubuntu 12.04 LTS. It seems that everything out there has
issues. I'm trying to get to the place where I can say, "Run these
commands and you're sure to have working (and up to date) .debs."

When I get an exact list of commands put together, I'll be sure to
update the wiki. The build process can be slow and I'm just looking to
save some time. Again, if you have anything, let me know.


Best,
Gabe


From ssinyagin at yahoo.com  Sat Sep 22 02:01:28 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Fri, 21 Sep 2012 15:01:28 -0700 (PDT)
Subject: [Freeswitch-users] Conference bridge example
Message-ID: <1348264888.61738.YahooMailNeo@web39306.mail.mud.yahoo.com>

I wrote a short tutorial based on my production server configuration. I guess it might be useful:
http://txlab.wordpress.com/2012/09/17/setting-up-a-conference-bridge-with-freeswitch/


cheers
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From cmason at frontiernetworks.ca  Sat Sep 22 03:33:24 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Fri, 21 Sep 2012 19:33:24 -0400
Subject: [Freeswitch-users] FreeSWITCH HA Database changes in 1.2
Message-ID: <0D1C698866F66045A6201FD0F59CAC900146996AD5@EX.frontier.local>

Hello,

I noticed recently on FS 1.2 that the table "sip_recovery" wasn't being created automatically by FreeSWITCH. If I specify:

<param name="core-recovery-db-dsn" value="maxpowersoft_odbc:freeswitch:blahblah"/>

In switch.conf.xml, FreeSWITCH will create a table called "recovery".



So my questions are:


1.       Have you renamed the sip_recovery table to recovery?

2.       Is "core-recovery-db-dsn" a required parameter for HA FreeSWITCH to function?

Thanks,
Colin
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From krice at freeswitch.org  Sat Sep 22 06:38:56 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 21 Sep 2012 21:38:56 -0500
Subject: [Freeswitch-users] just a link for you guys
Message-ID: <BA39A0A9-D09C-43FF-9FF1-8EF7C33B244D@freeswitch.org>

http://blog.krisk.org/2012/09/freeswitch-stable-branch-sponsored-by.html

If you don't know Kris, he is the original creator of Astlinux... check out his blog post.

And, now I can say thanks to Kris and the rest of the Star2Star crew for sponsoring the FreeSWITCH stable branch and me personally. Without companies like Star2Star, I wouldn't be working on FreeSWITCH full time and several other full time developers would be part time developers!

Thanks to all and read Kris' blog.

Ken
Sent from my iPad

From gabe at gundy.org  Sat Sep 22 08:51:54 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 21 Sep 2012 22:51:54 -0600
Subject: [Freeswitch-users] Mod xml curl and mod dingaling
In-Reply-To: <CACSMF4wUjNMat+N6f3MpUzJty=hgpOzoV2Bg2Sp069FcX1w87A@mail.gmail.com>
References: <CACSMF4wUjNMat+N6f3MpUzJty=hgpOzoV2Bg2Sp069FcX1w87A@mail.gmail.com>
Message-ID: <CAMwZ-tLc_EsigfT2nJdkasnL+HAHfHOVZpnKqFxd-oKcQoxJRw@mail.gmail.com>

Peter,

It's hard to know exactly what you're asking, so I'm going to ask a
few follow-up questions...

On Tue, Sep 18, 2012 at 4:57 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
> I'm setting up FreeSwitch and I'm using mod xml curl.
>
> I catch all calls and redirect them (post from Freeswitch to my servlet) to
> my handler.

OK, so we're talking about handeling the dialplan section of the XML here.



> I give all 'sip' extension configurations dynamically back in xml.

And now, we're talking about handeling the directory here, right?



> How does the complete configuration look of a basic Gtalk client?
>
> http://wiki.freeswitch.org/wiki/Dingaling#Sample_Configuration_.28Google.29

I think that link has the answer to your question. Doesn't it?


> Can you include the extension in the client profile?

You might be thinking about it incorrectly. From the looks of it (and
I don't do much with mod_dingaling), you don't tie a google account to
each user. It's more like you configure that google user in the same
way you might configure a sofia gateway -- it allows you to call into
and out of the system.


> Ps, is it also possible to stop automatic posts to my handler? I want to be
> able to control this manually by runing for example "reloadacl" command.

Nope. If you set a section of the XML to be dynamic, it's going to be
dynamic. You can however return a "not found" response and FreeSWITCH
will then look on the disk to see if there is anything available:

http://wiki.freeswitch.org/wiki/Mod_xml_curl#General_.22Not_found.22_reply


Good luck. Hope I helped more than I confused the situation :)



Gabe


From gabe at gundy.org  Sat Sep 22 08:59:53 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 21 Sep 2012 22:59:53 -0600
Subject: [Freeswitch-users] issue with web server to handle XML CDRs
In-Reply-To: <CAHqn9HFfo9JiB5mz747Pg18q5hPZXgfV6cnroyQF_SqBkcepzA@mail.gmail.com>
References: <CAHqn9HFfo9JiB5mz747Pg18q5hPZXgfV6cnroyQF_SqBkcepzA@mail.gmail.com>
Message-ID: <CAMwZ-t++e-M-fFYst1yNx+M0EOW0W8CX_Z9JFLNZb2qJXT89jQ@mail.gmail.com>

On Mon, Sep 17, 2012 at 3:36 AM, Jagadish Thoutam
<jaganthoutam at gmail.com> wrote:
> There is the issue with xml_cdr for me when i try to call <param name="url"
> value="http://localhost/cgi-bin/cdr.pl"/> its not working, i have load the
> mod_xml_cdr on cli, and my cdr.pl file is working well when try it manually
> and also have executable permission for cdr.pl file, please some one tell me
> what i am missing here.

>From your question, it sounds like you're not sure what side the
problem is on, the HTTP client (FreeSWITCH) or the HTTP server (your
cdr.pl).

We can't tell you where to look if we don't have more information.
Let's start by seeing if the HTTP clint (FreeSWITCH) is generating the
HTTP POST.  Do you see traffic on port 80? Try something like this:
ngrep -qt -W byline port 80. If you don't see it, the problem is your
FreeSWITCH config, if you do see it, something is messed up on the
webserver.

Hope that helps. Please let us know what you find.


Gabe


From gabe at gundy.org  Sat Sep 22 10:40:27 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 22 Sep 2012 00:40:27 -0600
Subject: [Freeswitch-users] Change the DID
In-Reply-To: <fed4e1549c758c9ef7b3069b07d4cb2d.squirrel@emailmg.globat.com>
References: <fed4e1549c758c9ef7b3069b07d4cb2d.squirrel@emailmg.globat.com>
Message-ID: <CAMwZ-tLsuJcqDDGZK_mnXmqOGYfM-YZg4Op6bjJZ3YrZZkZs0Q@mail.gmail.com>

On Fri, Sep 21, 2012 at 10:14 AM,  <rsaavedra at ecogizmos.com> wrote:
> How can I specified a different value for the DID?

Use these variables to alter the Caller ID:
http://wiki.freeswitch.org/wiki/Channel_Variables#Caller_ID_Related


Gabe


From peetzer at gmail.com  Sat Sep 22 10:33:38 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Sat, 22 Sep 2012 08:33:38 +0200
Subject: [Freeswitch-users] Mod xml curl and mod dingaling
In-Reply-To: <CAMwZ-tLc_EsigfT2nJdkasnL+HAHfHOVZpnKqFxd-oKcQoxJRw@mail.gmail.com>
References: <CACSMF4wUjNMat+N6f3MpUzJty=hgpOzoV2Bg2Sp069FcX1w87A@mail.gmail.com>
	<CAMwZ-tLc_EsigfT2nJdkasnL+HAHfHOVZpnKqFxd-oKcQoxJRw@mail.gmail.com>
Message-ID: <9F2EA28A-0156-4FE5-9F34-266622B182CB@gmail.com>

Hi Gabe,


On 22 sep 2012, at 06:51, Gabriel Gunderson wrote:

> Peter,
> 
> It's hard to know exactly what you're asking, so I'm going to ask a
> few follow-up questions...
> 
> On Tue, Sep 18, 2012 at 4:57 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
>> I'm setting up FreeSwitch and I'm using mod xml curl.
>> 
>> I catch all calls and redirect them (post from Freeswitch to my servlet) to
>> my handler.
> 
> OK, so we're talking about handeling the dialplan section of the XML here.
> 
> Yes, correct, I put section in the dialplan to my servlet.
> 
>> I give all 'sip' extension configurations dynamically back in xml.
> 
> And now, we're talking about handeling the directory here, right?
> 
> Yes only the directory (but can also be dialplan or configuration)
> 
>> How does the complete configuration look of a basic Gtalk client?
>> 
>> http://wiki.freeswitch.org/wiki/Dingaling#Sample_Configuration_.28Google.29
> 
> I think that link has the answer to your question. Doesn't it?
> 
This I have seen, but was wondering to put all together as with normal extension (directory). The config file is also referencing the exten (extension file) Since the xml is not static, I get the login/ password and type of phone from my application.
FreeSwitch asks me for the configuration by a post and I return the xml. This works for normal SIP extensions, just returning the xml with userid. I would like to return the dingaling and extension all together. 
> 
>> Can you include the extension in the client profile?
> 
> You might be thinking about it incorrectly. From the looks of it (and
> I don't do much with mod_dingaling), you don't tie a google account to
> each user. It's more like you configure that google user in the same
> way you might configure a sofia gateway -- it allows you to call into
> and out of the system.
> 
Only for Gtalk phones, I know the type.
> 
>> Ps, is it also possible to stop automatic posts to my handler? I want to be
>> able to control this manually by runing for example "reloadacl" command.
> 
> Nope. If you set a section of the XML to be dynamic, it's going to be
> dynamic. You can however return a "not found" response and FreeSWITCH
> will then look on the disk to see if there is anything available:
> 
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#General_.22Not_found.22_reply
> 
Numbers unknown or a specific alfanumeric extension I handle with fifo but this leads to a new question ;)
> 
> Good luck. Hope I helped more than I confused the situation :)
> 
> Thank you for replying and thinking along! The static configuration works with both Gtalk config and extension, this I tested. I want however to register all type of phones dynamically by mod curl and won't make use of static files. I have to return a Gtalk configuration + extension (user) somehow in my complete xml that I return.

http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
>  
> 
> 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 peetzer at gmail.com  Sat Sep 22 11:03:24 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Sat, 22 Sep 2012 09:03:24 +0200
Subject: [Freeswitch-users] Fifo api command register agents (add
	fifo_member)
Message-ID: <9205AA1A-DBDE-4B0F-92B6-37605C4A485E@gmail.com>

Hi all,

I started to implement fifo in my application to handle calls from know phones to unknown extensions and specific dial string/ extension.

I use the api commands with esl (java client), outbound socket (with sendMsg).

http://wiki.freeswitch.org/wiki/Mod_event_socket

I looked at a lot of examples, for me I thought the right sequence to would be to add (register) user agents first (determined by myself) and later add caller in which all user agents would ring and the call would be handled after accepting the call from free user agent.

My problem is now the right notation of the add fifo_member. I think I should use this for registering agents;

http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback

http://wiki.freeswitch.org/wiki/Mod_fifo#Simple%20On-hook%20Agent%20Login/Logout%20Example

http://wiki.freeswitch.org/wiki/Variable_fifo_member_wait

sendmsg <uuid>
call-command: execute
execute-app-name: <one of the applications>
execute-app-arg: <application data>
sendMsg
call-command: execute
execute-app-name: set
execute-app-arg:
"fifo_member(add TestQue sofia/internal/1001%123.123.123.123)"
I don't know if the () can be used in sendMsg, my queue name needs quotes "\'" and the callerid needs "@" or "%" with domain at end.

When I execute the command I don't get errors, response is ok and in log of freeswitch I also see no errors (only set command).

However when I execute fifo list or (verbose_list) from FS CLI, I don't see any user agents registered. If I put the caller in my queue, I see only the caller registered (callers) http://wiki.freeswitch.org/wiki/Mod_fifo#Terminology

http://wiki.freeswitch.org/wiki/Mod_fifo#list.7Clist_verbose

Below works, only my created queue is missing lag, call timeout and simcount!

Put a caller into a FIFO queue

sendMsg
call-command: execute
execute-app-name: fifo
execute-app-arg: TestQue in somesoundfile.wav someothersoundfilemusic.wav
Kind regards,

Peter

Ps, I hope somebody can advise !!!!
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From dujinfang at gmail.com  Sat Sep 22 15:31:38 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 22 Sep 2012 19:31:38 +0800
Subject: [Freeswitch-users] Fifo api command register agents (add
 fifo_member)
In-Reply-To: <9205AA1A-DBDE-4B0F-92B6-37605C4A485E@gmail.com>
References: <9205AA1A-DBDE-4B0F-92B6-37605C4A485E@gmail.com>
Message-ID: <1CD16CE835DC4C7C815E8D8599D81DF3@gmail.com>

sendmsg is not for api, try  


api fifo_member add blah????\n\n



--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Saturday, September 22, 2012 at 3:03 PM, Peter van Raamsdonk wrote:

> Hi all,
>  
> I started to implement fifo in my application to handle calls from know phones to unknown extensions and specific dial string/ extension.
>  
> I use the api commands with esl (java client), outbound socket (with sendMsg).
>  
> http://wiki.freeswitch.org/wiki/Mod_event_socket
>  
> I looked at a lot of examples, for me I thought the right sequence to would be to add (register) user agents first (determined by myself) and later add caller in which all user agents would ring and the call would be handled after accepting the call from free user agent.
>  
> My problem is now the right notation of the add fifo_member. I think I should use this for registering agents;
>  
> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>  
> http://wiki.freeswitch.org/wiki/Mod_fifo#Simple%20On-hook%20Agent%20Login/Logout%20Example
>  
> http://wiki.freeswitch.org/wiki/Variable_fifo_member_wait
>  
> sendmsg <uuid> call-command: execute execute-app-name: <one of the applications> execute-app-arg: <application data>
> sendMsg
> call-command: execute
> execute-app-name: set
> execute-app-arg:
> "fifo_member(add TestQue sofia/internal/1001%123.123.123.123)"
> I don't know if the () can be used in sendMsg, my queue name needs quotes "\'" and the callerid needs "@" or "%" with domain at end.
>  
>  
>  
>  
> When I execute the command I don't get errors, response is ok and in log of freeswitch I also see no errors (only set command).
>  
> However when I execute fifo list or (verbose_list) from FS CLI, I don't see any user agents registered. If I put the caller in my queue, I see only the caller registered (callers) http://wiki.freeswitch.org/wiki/Mod_fifo#Terminology
>  
> http://wiki.freeswitch.org/wiki/Mod_fifo#list.7Clist_verbose
>  
> Below works, only my created queue is missing lag, call timeout and simcount!
>  
> Put a caller into a FIFO queue
> sendMsg
> call-command: execute
> execute-app-name: fifo
> execute-app-arg: TestQue in somesoundfile.wav someothersoundfilemusic.wav
>  
> Kind regards,
>  
>  
>  
> Peter
>  
> Ps, I hope somebody can advise !!!!
> _________________________________________________________________________
> 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 b2m at a-cti.com  Sat Sep 22 16:03:17 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Sat, 22 Sep 2012 17:33:17 +0530
Subject: [Freeswitch-users] FreeSWITCH : mod_skypopen
In-Reply-To: <CALXCt0rHDtcv=4YQTqd5xpQS20oei=H6Bdn6=CP-TtUMHgBNHg@mail.gmail.com>
References: <CAP2fbbda3qpPOhJ+kUNU6B-F3MxwDZ6ebeBW3w1ZnPwBD4yngQ@mail.gmail.com>
	<CALXCt0rHDtcv=4YQTqd5xpQS20oei=H6Bdn6=CP-TtUMHgBNHg@mail.gmail.com>
Message-ID: <CAP2fbbfCNec1CDc2eFDzRbhtPpvwz5a+LRi6jub73UJuCH9igw@mail.gmail.com>

Thank you so much, going to try it.

Thanks,
Bala

On Fri, Sep 21, 2012 at 7:54 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> You can use the chatplan for chat (load mod_sms for this) and from
> chatplan you can use lua.
>
> Presence is not managed at the moment.
>
> If you are thinking at the presence of "others" (eg: the buddy list)
> this is not managed at the moment.
>
> If you are thinking at the presence of himself (eg: changing the
> presence of the interface from online to busy or away or do not
> disturb) it's not directly managed (it's forced to "online" at
> startup), but you can do that by sending commands to the skype client:
>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#skypopen
> "SET USERSTATUS ONLINE"
>
> from http://developer.skype.com/public-api-reference :
> GET USERSTATUS
>
> This command queries or modifies user visiblity for the current user.
>
> Syntax
> GET USERSTATUS
> SET USERSTATUS <value>
>
> Response
> USERSTATUS <value>
>
> Parameters
> <value> ? new userstatus. Possible values:
>
> UNKNOWN
> ONLINE ? current user is online
> OFFLINE ? current user is offline
> SKYPEME ? current user is in ?Skype Me? mode (protocol 2).
> AWAY ? current user is away.
> NA ? current user is not available.
> DND ? current user is in ?Do not disturb? mode.
> INVISIBLE ? current user is invisible to others.
> LOGGEDOUT ? current user is logged out. Clients are detached.
>
> Version
> Protocol 1
>
> Errors
>
> ERROR 28 Unknown userstatus
> Status value is incorrect or misspelled
>
> Example
>
> -> SET USERSTATUS OFFLINE
> <- USERSTATUS OFFLINE
> <- USERSTATUS OFFLINE
> -> SET USERSTATUS xxx
> <- ERROR 28 Unknown userstatus
>
>
>
> On Fri, Sep 21, 2012 at 4:10 PM, Balamurugan Mahendran <b2m at a-cti.com>
> wrote:
> >
> > Is it possible to handle Skype chat and presence through lua script?
> >
> > Thanks,
> > Bala
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 peetzer at gmail.com  Sat Sep 22 19:41:36 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Sat, 22 Sep 2012 17:41:36 +0200
Subject: [Freeswitch-users] Fifo api command register agents (add
	fifo_member)
In-Reply-To: <1CD16CE835DC4C7C815E8D8599D81DF3@gmail.com>
References: <9205AA1A-DBDE-4B0F-92B6-37605C4A485E@gmail.com>
	<1CD16CE835DC4C7C815E8D8599D81DF3@gmail.com>
Message-ID: <F5857C8A-7B8A-41BC-ACE5-AFAAF4B8FBE9@gmail.com>

Hi Seven,

Inbound and outboud is confusing because both can do the same sometime, I will try but this should also be possible with sendMsg.

Thanks,

Peter

On 22 sep 2012, at 13:31, Seven Du wrote:

> sendmsg is not for api, try
> 
> 
> api fifo_member add blah????\n\n
> 
> 
> 
> -- 
> Seven Du
> Sent with Sparrow
> 
> On Saturday, September 22, 2012 at 3:03 PM, Peter van Raamsdonk wrote:
> 
>> Hi all,
>> 
>> I started to implement fifo in my application to handle calls from know phones to unknown extensions and specific dial string/ extension.
>> 
>> I use the api commands with esl (java client), outbound socket (with sendMsg).
>> 
>> http://wiki.freeswitch.org/wiki/Mod_event_socket
>> 
>> I looked at a lot of examples, for me I thought the right sequence to would be to add (register) user agents first (determined by myself) and later add caller in which all user agents would ring and the call would be handled after accepting the call from free user agent.
>> 
>> My problem is now the right notation of the add fifo_member. I think I should use this for registering agents;
>> 
>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>> 
>> http://wiki.freeswitch.org/wiki/Mod_fifo#Simple%20On-hook%20Agent%20Login/Logout%20Example
>> 
>> http://wiki.freeswitch.org/wiki/Variable_fifo_member_wait
>> 
>> sendmsg <uuid>
>> call-command: execute
>> execute-app-name: <one of the applications>
>> execute-app-arg: <application data>
>> sendMsg
>> call-command: execute
>> execute-app-name: set
>> execute-app-arg:
>> "fifo_member(add TestQue sofia/internal/1001%123.123.123.123)"
>> I don't know if the () can be used in sendMsg, my queue name needs quotes "\'" and the callerid needs "@" or "%" with domain at end.
>> 
>> When I execute the command I don't get errors, response is ok and in log of freeswitch I also see no errors (only set command).
>> 
>> However when I execute fifo list or (verbose_list) from FS CLI, I don't see any user agents registered. If I put the caller in my queue, I see only the caller registered (callers) http://wiki.freeswitch.org/wiki/Mod_fifo#Terminology
>> 
>> http://wiki.freeswitch.org/wiki/Mod_fifo#list.7Clist_verbose
>> 
>> Below works, only my created queue is missing lag, call timeout and simcount!
>> 
>> Put a caller into a FIFO queue
>> 
>> sendMsg
>> call-command: execute
>> execute-app-name: fifo
>> execute-app-arg: TestQue in somesoundfile.wav someothersoundfilemusic.wav
>> Kind regards,
>> 
>> Peter
>> 
>> Ps, I hope somebody can advise !!!!
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Sep 22 22:08:49 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 23 Sep 2012 02:08:49 +0800
Subject: [Freeswitch-users] Fifo api command register agents (add
 fifo_member)
In-Reply-To: <F5857C8A-7B8A-41BC-ACE5-AFAAF4B8FBE9@gmail.com>
References: <9205AA1A-DBDE-4B0F-92B6-37605C4A485E@gmail.com>
	<1CD16CE835DC4C7C815E8D8599D81DF3@gmail.com>
	<F5857C8A-7B8A-41BC-ACE5-AFAAF4B8FBE9@gmail.com>
Message-ID: <E5CAC22881994A6891B9607305A6DF10@gmail.com>

It should work with inbound and outbound, but if you use sendMsg, you need a session.  

--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Saturday, September 22, 2012 at 11:41 PM, Peter van Raamsdonk wrote:

> Hi Seven,
>  
> Inbound and outboud is confusing because both can do the same sometime, I will try but this should also be possible with sendMsg.
>  
> Thanks,
>  
> Peter
> On 22 sep 2012, at 13:31, Seven Du wrote:
> > sendmsg is not for api, try  
> >  
> >  
> > api fifo_member add blah????\n\n
> >  
> >  
> >  
> > --  
> > Seven Du
> > Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
> >  
> >  
> > On Saturday, September 22, 2012 at 3:03 PM, Peter van Raamsdonk wrote:
> >  
> > > Hi all,
> > >  
> > > I started to implement fifo in my application to handle calls from know phones to unknown extensions and specific dial string/ extension.
> > >  
> > > I use the api commands with esl (java client), outbound socket (with sendMsg).
> > >  
> > > http://wiki.freeswitch.org/wiki/Mod_event_socket
> > >  
> > > I looked at a lot of examples, for me I thought the right sequence to would be to add (register) user agents first (determined by myself) and later add caller in which all user agents would ring and the call would be handled after accepting the call from free user agent.
> > >  
> > > My problem is now the right notation of the add fifo_member. I think I should use this for registering agents;
> > >  
> > > http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
> > >  
> > > http://wiki.freeswitch.org/wiki/Mod_fifo#Simple%20On-hook%20Agent%20Login/Logout%20Example
> > >  
> > > http://wiki.freeswitch.org/wiki/Variable_fifo_member_wait
> > >  
> > > sendmsg <uuid> call-command: execute execute-app-name: <one of the applications> execute-app-arg: <application data>
> > > sendMsg
> > > call-command: execute
> > > execute-app-name: set
> > > execute-app-arg:
> > > "fifo_member(add TestQue sofia/internal/1001%123.123.123.123)"
> > > I don't know if the () can be used in sendMsg, my queue name needs quotes "\'" and the callerid needs "@" or "%" with domain at end.
> > >  
> > >  
> > >  
> > >  
> > > When I execute the command I don't get errors, response is ok and in log of freeswitch I also see no errors (only set command).
> > >  
> > > However when I execute fifo list or (verbose_list) from FS CLI, I don't see any user agents registered. If I put the caller in my queue, I see only the caller registered (callers) http://wiki.freeswitch.org/wiki/Mod_fifo#Terminology
> > >  
> > > http://wiki.freeswitch.org/wiki/Mod_fifo#list.7Clist_verbose
> > >  
> > > Below works, only my created queue is missing lag, call timeout and simcount!
> > >  
> > > Put a caller into a FIFO queue
> > > sendMsg
> > > call-command: execute
> > > execute-app-name: fifo
> > > execute-app-arg: TestQue in somesoundfile.wav someothersoundfilemusic.wav
> > >  
> > > Kind regards,
> > >  
> > >  
> > >  
> > > Peter
> > >  
> > > Ps, I hope somebody can advise !!!!
> > > _________________________________________________________________________
> > > 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
> >  
> > 
> > 
> >  
> > Official 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 gabe at gundy.org  Sun Sep 23 05:30:56 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 22 Sep 2012 19:30:56 -0600
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
Message-ID: <CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>

On Fri, Sep 14, 2012 at 11:54 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
> I tried the mod_xml_curl after studying the wiki. It works like a charm and
> FS post to my java servlet easily (dialplan and user directory).
> I read on a blog there is option 'cacheable=true' to prevent a post every
> time a dial is made, do you know where to put this?

I've never seen this. I know there was some talk about it (I can't
remember if it was ClueCon or the mailing list or what). But I think
the general consensus was that simple cacheing isn't going to do much
good. It's almost like you need to know, based on the POST info if you
should cache or not.

I'd love a link to the blog if you can find it.  Meanwhile, I'm
running off to the source to see if that's true.

If it's something you really need, I'd consider adding caching at the
webserver layer, take the work of evaluating the request away from
freeswitch and move it to another box.

Also, you could write your own dialplan that worked as a caching
layer. First FreeSWITCH hits your custom dialplan, if it returns a
cached dialplan you're done. If not, it moves on to mod_curl_xml. And
finally, it makes its way to disk if it can't find a suitable
dialplan. I've never done it, but I hear it's not that hard to write.


Good luck.
Gabe


From mike.burlingame at me.com  Sun Sep 23 06:46:38 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sat, 22 Sep 2012 19:46:38 -0700
Subject: [Freeswitch-users] FreeSwitch Video Calls Bria iPhone Client
Message-ID: <2B4CD419-FE1B-443C-86AB-2E5A98C63ED2@me.com>

I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android? 

Thanks



From mike.burlingame at me.com  Sun Sep 23 07:02:18 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sat, 22 Sep 2012 20:02:18 -0700
Subject: [Freeswitch-users] FreeSwitch not Processing calls
 (B-Leg)	until restart
In-Reply-To: <AD377EE5-7A7F-438A-9E0F-ADE2936D9DFE@me.com>
References: <213A77A7-64E9-4BA8-AF67-13E12FC86927@me.com>
	<CAKbxfG-eD63rN7f=D=-tz8nUDdWb0A=L7k6_NDOYnFOWZRzjvg@mail.gmail.com>
	<AD377EE5-7A7F-438A-9E0F-ADE2936D9DFE@me.com>
Message-ID: <5E9FC1AA-FD81-4292-A94F-016DE126809E@me.com>

So this happened again tried the fs_cli command same thing it froze tried to get a core dump using gore

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND                                                                                                                                                     
21212 root      -2 -10  511m  81m 6832 S    2  4.1   6:37.32 freeswitch

gcore 21212
/tmp/core.uQm660:1: Error in sourced command file:
/usr/local/freeswitch/bin/freeswitch (deleted): No such file or directory.
gcore: failed to create core.21212

 ls /usr/local/freeswitch/bin/
freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav


On Sep 16, 2012, at 6:59 PM, Mike Burlingame <mike.burlingame at me.com> wrote:

> Noted 
> 
> Thank you
> On Sep 16, 2012, at 6:56 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> recourse in that situation would be to try
>> 
>> fs_cli -x "<cmd>"
>> 
>> if all else fails, gcore the fs process with the gcore command and get a bt just like a crash.
>> 
>> 
>> On Sun, Sep 16, 2012 at 8:21 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> I had an issue today not sure if anyone else has seen this and would be willing to share some information if you have - Today FreeSwitch stopped processing calls correctly the A-Leg of the call would come into FreeSwitch -  i see the notice for the new channel however no sip signaling was sent out for the B-Leg - I could not do any debug because fs_cli was locked up, force quoting fs_cli and restarting it yielded seeing the sip signaling however as soon as I type anything at the prompt fs_cli would lock again. The only resolution was to restart the FreeSwitch process and everything worked as expected again.
>> 
>> Has anyone else seen anything like this using git's from Sept 14th?
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 paul at cupis.co.uk  Sun Sep 23 12:05:59 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Sun, 23 Sep 2012 09:05:59 +0100
Subject: [Freeswitch-users] Newbie question,
 FS cli and db (users for user directory)
In-Reply-To: <CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
Message-ID: <505EC2E7.6070700@cupis.co.uk>

On 23/09/12 02:30, Gabriel Gunderson wrote:
> On Fri, Sep 14, 2012 at 11:54 AM, Peter van Raamsdonk<peetzer at gmail.com>  wrote:
>> I tried the mod_xml_curl after studying the wiki. It works like a charm and
>> FS post to my java servlet easily (dialplan and user directory).
>> I read on a blog there is option 'cacheable=true' to prevent a post every
>> time a dial is made, do you know where to put this?

> I'd love a link to the blog if you can find it.  Meanwhile, I'm
> running off to the source to see if that's true.

Possibly:

http://blog.godson.in/2011/06/pitfalls-to-avoid-while-using.html

Regards,


From william.king at quentustech.com  Sun Sep 23 12:17:25 2012
From: william.king at quentustech.com (William King)
Date: Sun, 23 Sep 2012 01:17:25 -0700
Subject: [Freeswitch-users] Building v1.2 .debs on Ubuntu 12.04 LTS
In-Reply-To: <CAMwZ-t+bqZ6NUc5f7z2hFB3dFcNpT=86v3kfH_62Ws=M_Rs6yg@mail.gmail.com>
References: <CAMwZ-t+bqZ6NUc5f7z2hFB3dFcNpT=86v3kfH_62Ws=M_Rs6yg@mail.gmail.com>
Message-ID: <505EC595.8010700@quentustech.com>

There are scripts for building the dsc and deb files in tree right now.
check the ./scripts/ci/ folder.
There has been a lot of work recently to get a nightly and a stable repo
that has the debian and ubuntu packages. So that is something to keep an
eye out for.

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 09/21/2012 02:09 PM, Gabriel Gunderson wrote:
> All,
>
> Before I get started down this path again, I'd like to know if anyone
> has a clean list of commands to build FreeSWITCH 1.2 debs on a clean
> install of Ubuntu 12.04 LTS. It seems that everything out there has
> issues. I'm trying to get to the place where I can say, "Run these
> commands and you're sure to have working (and up to date) .debs."
>
> When I get an exact list of commands put together, I'll be sure to
> update the wiki. The build process can be slow and I'm just looking to
> save some time. Again, if you have anything, 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sun Sep 23 17:10:28 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 23 Sep 2012 08:10:28 -0500
Subject: [Freeswitch-users] FreeSwitch not Processing calls (B-Leg)
	until restart
In-Reply-To: <5E9FC1AA-FD81-4292-A94F-016DE126809E@me.com>
References: <213A77A7-64E9-4BA8-AF67-13E12FC86927@me.com>
	<CAKbxfG-eD63rN7f=D=-tz8nUDdWb0A=L7k6_NDOYnFOWZRzjvg@mail.gmail.com>
	<AD377EE5-7A7F-438A-9E0F-ADE2936D9DFE@me.com>
	<5E9FC1AA-FD81-4292-A94F-016DE126809E@me.com>
Message-ID: <CAKbxfG_xQyXTp1ZOoHf-hxDX=v2si40Vj6SVMADMBCA5r1qtOA@mail.gmail.com>

Did you maybe make current while it was running or something?  Sounds like
you nuked your install dir perhaps?
On Sep 22, 2012 10:03 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:

> So this happened again tried the fs_cli command same thing it froze tried
> to get a core dump using gore
>
>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>
>
> 21212 root      -2 -10  511m  81m 6832 S    2  4.1   6:37.32 freeswitch
>
> gcore 21212
> /tmp/core.uQm660:1: Error in sourced command file:
> /usr/local/freeswitch/bin/freeswitch (deleted): No such file or directory.
> gcore: failed to create core.21212
>
>  ls /usr/local/freeswitch/bin/
> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav
>
>
> On Sep 16, 2012, at 6:59 PM, Mike Burlingame <mike.burlingame at me.com>
> wrote:
>
> Noted
>
> Thank you
> On Sep 16, 2012, at 6:56 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> recourse in that situation would be to try
>
> fs_cli -x "<cmd>"
>
> if all else fails, gcore the fs process with the gcore command and get a
> bt just like a crash.
>
>
> On Sun, Sep 16, 2012 at 8:21 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> I had an issue today not sure if anyone else has seen this and would be
>> willing to share some information if you have - Today FreeSwitch stopped
>> processing calls correctly the A-Leg of the call would come into FreeSwitch
>> -  i see the notice for the new channel however no sip signaling was sent
>> out for the B-Leg - I could not do any debug because fs_cli was locked up,
>> force quoting fs_cli and restarting it yielded seeing the sip signaling
>> however as soon as I type anything at the prompt fs_cli would lock again.
>> The only resolution was to restart the FreeSwitch process and everything
>> worked as expected again.
>>
>> Has anyone else seen anything like this using git's from Sept 14th?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
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From mike.burlingame at me.com  Sun Sep 23 18:46:43 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sun, 23 Sep 2012 07:46:43 -0700
Subject: [Freeswitch-users] FreeSwitch not Processing calls
 (B-Leg)	until restart
In-Reply-To: <CAKbxfG_xQyXTp1ZOoHf-hxDX=v2si40Vj6SVMADMBCA5r1qtOA@mail.gmail.com>
References: <213A77A7-64E9-4BA8-AF67-13E12FC86927@me.com>
	<CAKbxfG-eD63rN7f=D=-tz8nUDdWb0A=L7k6_NDOYnFOWZRzjvg@mail.gmail.com>
	<AD377EE5-7A7F-438A-9E0F-ADE2936D9DFE@me.com>
	<5E9FC1AA-FD81-4292-A94F-016DE126809E@me.com>
	<CAKbxfG_xQyXTp1ZOoHf-hxDX=v2si40Vj6SVMADMBCA5r1qtOA@mail.gmail.com>
Message-ID: <DE515039-A8F2-4216-9641-F7B85C8DEF55@me.com>

No - no changes has been made to the server AFAIK - I will restart the process and revoke all SSH keys just to be sure someone else has not been in the server and made and changes

Sent from my iPhone 4S

On Sep 23, 2012, at 6:10 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Did you maybe make current while it was running or something?  Sounds like you nuked your install dir perhaps?
> 
> On Sep 22, 2012 10:03 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
>> So this happened again tried the fs_cli command same thing it froze tried to get a core dump using gore
>> 
>>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND                                                                                                                                                    
>> 21212 root      -2 -10  511m  81m 6832 S    2  4.1   6:37.32 freeswitch
>> 
>> gcore 21212
>> /tmp/core.uQm660:1: Error in sourced command file:
>> /usr/local/freeswitch/bin/freeswitch (deleted): No such file or directory.
>> gcore: failed to create core.21212
>> 
>>  ls /usr/local/freeswitch/bin/
>> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav
>> 
>> 
>> On Sep 16, 2012, at 6:59 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> 
>>> Noted 
>>> 
>>> Thank you
>>> On Sep 16, 2012, at 6:56 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>> 
>>>> recourse in that situation would be to try
>>>> 
>>>> fs_cli -x "<cmd>"
>>>> 
>>>> if all else fails, gcore the fs process with the gcore command and get a bt just like a crash.
>>>> 
>>>> 
>>>> On Sun, Sep 16, 2012 at 8:21 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>>> I had an issue today not sure if anyone else has seen this and would be willing to share some information if you have - Today FreeSwitch stopped processing calls correctly the A-Leg of the call would come into FreeSwitch -  i see the notice for the new channel however no sip signaling was sent out for the B-Leg - I could not do any debug because fs_cli was locked up, force quoting fs_cli and restarting it yielded seeing the sip signaling however as soon as I type anything at the prompt fs_cli would lock again. The only resolution was to restart the FreeSwitch process and everything worked as expected again.
>>>>> 
>>>>> Has anyone else seen anything like this using git's from Sept 14th?
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From darcy at Vex.Net  Sun Sep 23 18:37:50 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Sun, 23 Sep 2012 10:37:50 -0400
Subject: [Freeswitch-users] Some questions from a relative newbie
Message-ID: <20120923103750.543c14ff1599ab5fbfc0427e@Vex.Net>

Hello.  Long time listener, first time caller.  I have a few questions
but I will split them up into different emails for the benefit of
future mailing list searches.  I just wanted to introduce myself first
and explain what I am trying to do in case someone has some general
advice for me.

I own and run a small ISP up here in Toronto, Canada.  One of my
offerings recently has been VOIP through another provider.  I have
decided to bite the bullet and have set up my own switch (Freeswitch,
of course) and connected it to an aggregator, VOIP Innovations.

I have a home grown billing system that I use to provision other parts
of my ISP and I am going to do the same with VOIP.  I have cron jobs
that check the database and, if necessary, rebuild certain
configuration files and reload them.  That works with the basic stuff I
am doing so far.

I am running on FreeBSD 9.0-RELEASE.  As I am also a NetBSD developer I
would like to move it over but FreeBSD has a port and NetSBD does not
yet have a package for Freeswitch.  I will probably be looking at
creating one eventually.

And here is my first question.  I have worked around this but I am
curious.  Has the behaviour of fs_cli under error conditions been
changed recently?  It seems to me that an earlier version worked with
FreeBSD's startup script but 1.2.3 fails.  The issue is a command
issued by the script:

  /usr/local/bin/fs_cli -x "sofia recover"

The problem is that the command is issued before the switch is actually
running.  I am not sure what good that is.  Since it fails, the startup
script doesn't complete and the switch never starts.  I have fixed the
script so that this command runs properly but I was wondering why the
switch even started up before.  Did fs_cli use to always return success
no matter what?

Cheers.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From darcy at Vex.Net  Sun Sep 23 19:01:21 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Sun, 23 Sep 2012 11:01:21 -0400
Subject: [Freeswitch-users] Keeping local calls local
Message-ID: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>

The first thing I found with the basic setup of FreeSWITCH was that
calls between my own clients still went to my provider.  This seemed
like a waste to me.  To handle calls coming in I generate an XML file
called dialplan/public/00_Local.xml.  Here is an example entry from it.

<extension name="Local_4164251212">
  <condition field="destination_number" expression="^(4164251212)$">
    <action application="export" data="dialed_extension=$1"/>
    <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
    <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
    <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
    <action application="set" data="ringback=${us-ring}"/>
    <action application="set" data="transfer_ringback=$${hold_music}"/>
    <action application="set" data="call_timeout=30"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="set" data="continue_on_fail=true"/>
    <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
    <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
    <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
    <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
    <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
    <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
    <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
    <action application="answer"/>
    <action application="sleep" data="1000"/>
    <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
  </condition>
</extension>

I am still trying to figure out exactly what everything in there does
but it seems to do the job.  Calls originating outside get delivered to
the correct client's phone.  Calls between clients go out to the
provider and then come back in as if it was external.

I worked around this by symlinking this file to the dialplan/default
directory.  This works but I wonder if this is the best way to
accomplish this.

Side question, can I create a simple alias for all of my DIDs and make
one condition that tests that alias somehow?

Cheers.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From peetzer at gmail.com  Sun Sep 23 19:05:53 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Sun, 23 Sep 2012 17:05:53 +0200
Subject: [Freeswitch-users] Fifo api command register agents (add
	fifo_member)
In-Reply-To: <E5CAC22881994A6891B9607305A6DF10@gmail.com>
References: <9205AA1A-DBDE-4B0F-92B6-37605C4A485E@gmail.com>
	<1CD16CE835DC4C7C815E8D8599D81DF3@gmail.com>
	<F5857C8A-7B8A-41BC-ACE5-AFAAF4B8FBE9@gmail.com>
	<E5CAC22881994A6891B9607305A6DF10@gmail.com>
Message-ID: <C84C9DB0-FBDF-4987-8CF0-3BB21AEF5A37@gmail.com>

I use the channel from the context and have the UUID.

Do you have any example how the command looks? It is hard to tell if it works, since the reply is +OK but I don't see the agent(s) registered.

I know the ${something} variables should be fetched separately, not parsed directly (well I assume). I don't know how the proper notation of functions with "(" and ")" should be.

Regards Peter

On 22 sep 2012, at 20:08, Seven Du wrote:

> It should work with inbound and outbound, but if you use sendMsg, you need a session.
> 
> -- 
> Seven Du
> Sent with Sparrow
> 
> On Saturday, September 22, 2012 at 11:41 PM, Peter van Raamsdonk wrote:
> 
>> Hi Seven,
>> 
>> Inbound and outboud is confusing because both can do the same sometime, I will try but this should also be possible with sendMsg.
>> 
>> Thanks,
>> 
>> Peter
>> 
>> On 22 sep 2012, at 13:31, Seven Du wrote:
>> 
>>> sendmsg is not for api, try
>>> 
>>> 
>>> api fifo_member add blah????\n\n
>>> 
>>> 
>>> 
>>> -- 
>>> Seven Du
>>> Sent with Sparrow
>>> 
>>> On Saturday, September 22, 2012 at 3:03 PM, Peter van Raamsdonk wrote:
>>> 
>>>> Hi all,
>>>> 
>>>> I started to implement fifo in my application to handle calls from know phones to unknown extensions and specific dial string/ extension.
>>>> 
>>>> I use the api commands with esl (java client), outbound socket (with sendMsg).
>>>> 
>>>> http://wiki.freeswitch.org/wiki/Mod_event_socket
>>>> 
>>>> I looked at a lot of examples, for me I thought the right sequence to would be to add (register) user agents first (determined by myself) and later add caller in which all user agents would ring and the call would be handled after accepting the call from free user agent.
>>>> 
>>>> My problem is now the right notation of the add fifo_member. I think I should use this for registering agents;
>>>> 
>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>> 
>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Simple%20On-hook%20Agent%20Login/Logout%20Example
>>>> 
>>>> http://wiki.freeswitch.org/wiki/Variable_fifo_member_wait
>>>> 
>>>> sendmsg <uuid>
>>>> call-command: execute
>>>> execute-app-name: <one of the applications>
>>>> execute-app-arg: <application data>
>>>> sendMsg
>>>> call-command: execute
>>>> execute-app-name: set
>>>> execute-app-arg:
>>>> "fifo_member(add TestQue sofia/internal/1001%123.123.123.123)"
>>>> I don't know if the () can be used in sendMsg, my queue name needs quotes "\'" and the callerid needs "@" or "%" with domain at end.
>>>> 
>>>> When I execute the command I don't get errors, response is ok and in log of freeswitch I also see no errors (only set command).
>>>> 
>>>> However when I execute fifo list or (verbose_list) from FS CLI, I don't see any user agents registered. If I put the caller in my queue, I see only the caller registered (callers) http://wiki.freeswitch.org/wiki/Mod_fifo#Terminology
>>>> 
>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#list.7Clist_verbose
>>>> 
>>>> Below works, only my created queue is missing lag, call timeout and simcount!
>>>> 
>>>> Put a caller into a FIFO queue
>>>> 
>>>> sendMsg
>>>> call-command: execute
>>>> execute-app-name: fifo
>>>> execute-app-arg: TestQue in somesoundfile.wav someothersoundfilemusic.wav
>>>> Kind regards,
>>>> 
>>>> Peter
>>>> 
>>>> Ps, I hope somebody can advise !!!!
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peetzer at gmail.com  Sun Sep 23 19:17:17 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Sun, 23 Sep 2012 17:17:17 +0200
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
Message-ID: <C43E3862-427B-454D-AA1C-F59AB3F0F391@gmail.com>

Hi Gabe,

The part although not the same website is on;

https://profiles.google.com/100965631603901259030/buzz

"Another tip is to use cacheable='true' attribute of user xml returned by web service. This will make FS to maintain a local cache of the account details. The details will stay in cache until cleared. This means that FS won't be hitting your web app for every REGISTER request that hoards of SIP UAs are sending, it will ask only once upon first request. The subsequent requests will be looked up from cache."

I'm trying to reduce unnecessary traffic (posts). The dialplan will probably don't change, only the initial servlet path I need to dynamically set. I can add this to my xml as variable and see if there is any change.

Thanks and have a good Sunday yet!

Peter

On 23 sep 2012, at 03:30, Gabriel Gunderson wrote:

> On Fri, Sep 14, 2012 at 11:54 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
>> I tried the mod_xml_curl after studying the wiki. It works like a charm and
>> FS post to my java servlet easily (dialplan and user directory).
>> I read on a blog there is option 'cacheable=true' to prevent a post every
>> time a dial is made, do you know where to put this?
> 
> I've never seen this. I know there was some talk about it (I can't
> remember if it was ClueCon or the mailing list or what). But I think
> the general consensus was that simple cacheing isn't going to do much
> good. It's almost like you need to know, based on the POST info if you
> should cache or not.
> 
> I'd love a link to the blog if you can find it.  Meanwhile, I'm
> running off to the source to see if that's true.
> 
> If it's something you really need, I'd consider adding caching at the
> webserver layer, take the work of evaluating the request away from
> freeswitch and move it to another box.
> 
> Also, you could write your own dialplan that worked as a caching
> layer. First FreeSWITCH hits your custom dialplan, if it returns a
> cached dialplan you're done. If not, it moves on to mod_curl_xml. And
> finally, it makes its way to disk if it can't find a suitable
> dialplan. I've never done it, but I hear it's not that hard to write.
> 
> 
> 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 peetzer at gmail.com  Sun Sep 23 19:19:02 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Sun, 23 Sep 2012 17:19:02 +0200
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <505EC2E7.6070700@cupis.co.uk>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
	<505EC2E7.6070700@cupis.co.uk>
Message-ID: <A987CDF3-7B2F-4412-A7F1-E08194026673@gmail.com>

Yep correct, that was the blog.

Regards Peter

On 23 sep 2012, at 10:05, Paul Cupis wrote:

> On 23/09/12 02:30, Gabriel Gunderson wrote:
>> On Fri, Sep 14, 2012 at 11:54 AM, Peter van Raamsdonk<peetzer at gmail.com>  wrote:
>>> I tried the mod_xml_curl after studying the wiki. It works like a charm and
>>> FS post to my java servlet easily (dialplan and user directory).
>>> I read on a blog there is option 'cacheable=true' to prevent a post every
>>> time a dial is made, do you know where to put this?
> 
>> I'd love a link to the blog if you can find it.  Meanwhile, I'm
>> running off to the source to see if that's true.
> 
> Possibly:
> 
> http://blog.godson.in/2011/06/pitfalls-to-avoid-while-using.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sun Sep 23 21:24:22 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 23 Sep 2012 10:24:22 -0700
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
Message-ID: <CAMWiMy8pSYSs5kPWm7Dh=a--A69PBz_m64KX8O3bdfg4Cc0YAg@mail.gmail.com>

You can use modules like mod_enum or mod_easyroute to route the calls
first before hitting your SIP provider. The wiki is a great place to
start for this type of scenario

On Sun, Sep 23, 2012 at 8:01 AM, D'Arcy Cain <darcy at vex.net> wrote:
> The first thing I found with the basic setup of FreeSWITCH was that
> calls between my own clients still went to my provider.  This seemed
> like a waste to me.  To handle calls coming in I generate an XML file
> called dialplan/public/00_Local.xml.  Here is an example entry from it.
>
> <extension name="Local_4164251212">
>   <condition field="destination_number" expression="^(4164251212)$">
>     <action application="export" data="dialed_extension=$1"/>
>     <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
>     <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>     <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
>     <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
>     <action application="set" data="ringback=${us-ring}"/>
>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>     <action application="set" data="call_timeout=30"/>
>     <action application="set" data="hangup_after_bridge=true"/>
>     <action application="set" data="continue_on_fail=true"/>
>     <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>     <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>     <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
>     <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>     <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>     <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>     <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>   </condition>
> </extension>
>
> I am still trying to figure out exactly what everything in there does
> but it seems to do the job.  Calls originating outside get delivered to
> the correct client's phone.  Calls between clients go out to the
> provider and then come back in as if it was external.
>
> I worked around this by symlinking this file to the dialplan/default
> directory.  This works but I wonder if this is the best way to
> accomplish this.
>
> Side question, can I create a simple alias for all of my DIDs and make
> one condition that tests that alias somehow?
>
> Cheers.
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ssinyagin at yahoo.com  Sun Sep 23 22:17:37 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
Message-ID: <1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>

here's a piece of my dialplan in "default" context. It simply catches such calls and transfers them directly to the public context. The public context matches local DID numbers and transfers them to corresponding extensions in default context:

??? <extension name="internal_did">
????? <condition field="destination_number" expression="^($${default_did_prefix}\d+)$">
??????? <action application="transfer" data="$1 XML public"/>
????? </condition>
??? </extension>


The "default_did_prefix" variable holds my DID number block. 






>________________________________
> From: D'Arcy Cain <darcy at Vex.Net>
>To: FreeSWITCH-users at lists.freeswitch.org 
>Sent: Sunday, September 23, 2012 5:01 PM
>Subject: [Freeswitch-users] Keeping local calls local
> 
>The first thing I found with the basic setup of FreeSWITCH was that
>calls between my own clients still went to my provider.? This seemed
>like a waste to me.? To handle calls coming in I generate an XML file
>called dialplan/public/00_Local.xml.? Here is an example entry from it.
>
><extension name="Local_4164251212">
>? <condition field="destination_number" expression="^(4164251212)$">
>? ? <action application="export" data="dialed_extension=$1"/>
>? ? <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
>? ? <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>? ? <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
>? ? <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
>? ? <action application="set" data="ringback=${us-ring}"/>
>? ? <action application="set" data="transfer_ringback=$${hold_music}"/>
>? ? <action application="set" data="call_timeout=30"/>
>? ? <action application="set" data="hangup_after_bridge=true"/>
>? ? <action application="set" data="continue_on_fail=true"/>
>? ? <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>? ? <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>? ? <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
>? ? <action application="answer"/>
>? ? <action application="sleep" data="1000"/>
>? ? <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>? </condition>
></extension>
>
>I am still trying to figure out exactly what everything in there does
>but it seems to do the job.? Calls originating outside get delivered to
>the correct client's phone.? Calls between clients go out to the
>provider and then come back in as if it was external.
>
>I worked around this by symlinking this file to the dialplan/default
>directory.? This works but I wonder if this is the best way to
>accomplish this.
>
>Side question, can I create a simple alias for all of my DIDs and make
>one condition that tests that alias somehow?
>
>Cheers.
>
>-- 
>D'Arcy J.M. Cain
>System Administrator, Vex.Net
>http://www.Vex.Net/ IM:darcy at Vex.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
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.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  Sun Sep 23 22:24:59 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sun, 23 Sep 2012 11:24:59 -0700 (PDT)
Subject: [Freeswitch-users] Some questions from a relative newbie
In-Reply-To: <20120923103750.543c14ff1599ab5fbfc0427e@Vex.Net>
References: <20120923103750.543c14ff1599ab5fbfc0427e@Vex.Net>
Message-ID: <1348424699.12793.YahooMailNeo@web39304.mail.mud.yahoo.com>

by the way, do you consider running a fault-tolerant service with OpenSIPS/Kamailio/Repro as a border controller and two FreeSWITCH boxes behind?




>________________________________
> From: D'Arcy Cain <darcy at Vex.Net>
>To: FreeSWITCH-users at lists.freeswitch.org 
>Sent: Sunday, September 23, 2012 4:37 PM
>Subject: [Freeswitch-users] Some questions from a relative newbie
> 
>Hello.? Long time listener, first time caller.? I have a few questions
>but I will split them up into different emails for the benefit of
>future mailing list searches.? I just wanted to introduce myself first
>and explain what I am trying to do in case someone has some general
>advice for me.
>
>I own and run a small ISP up here in Toronto, Canada.? One of my
>offerings recently has been VOIP through another provider.? I have
>decided to bite the bullet and have set up my own switch (Freeswitch,
>of course) and connected it to an aggregator, VOIP Innovations.
>
>I have a home grown billing system that I use to provision other parts
>of my ISP and I am going to do the same with VOIP.? I have cron jobs
>that check the database and, if necessary, rebuild certain
>configuration files and reload them.? That works with the basic stuff I
>am doing so far.
>
>I am running on FreeBSD 9.0-RELEASE.? As I am also a NetBSD developer I
>would like to move it over but FreeBSD has a port and NetSBD does not
>yet have a package for Freeswitch.? I will probably be looking at
>creating one eventually.
>
>And here is my first question.? I have worked around this but I am
>curious.? Has the behaviour of fs_cli under error conditions been
>changed recently?? It seems to me that an earlier version worked with
>FreeBSD's startup script but 1.2.3 fails.? The issue is a command
>issued by the script:
>
>? /usr/local/bin/fs_cli -x "sofia recover"
>
>The problem is that the command is issued before the switch is actually
>running.? I am not sure what good that is.? Since it fails, the startup
>script doesn't complete and the switch never starts.? I have fixed the
>script so that this command runs properly but I was wondering why the
>switch even started up before.? Did fs_cli use to always return success
>no matter what?
>
>Cheers.
>
>-- 
>D'Arcy J.M. Cain
>System Administrator, Vex.Net
>http://www.Vex.Net/ IM:darcy at Vex.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
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
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From darcy at Vex.Net  Sun Sep 23 23:10:07 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Sun, 23 Sep 2012 15:10:07 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <CAMWiMy8pSYSs5kPWm7Dh=a--A69PBz_m64KX8O3bdfg4Cc0YAg@mail.gmail.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<CAMWiMy8pSYSs5kPWm7Dh=a--A69PBz_m64KX8O3bdfg4Cc0YAg@mail.gmail.com>
Message-ID: <20120923151007.9194755c6f9b9979a3b2f4a2@Vex.Net>

On Sun, 23 Sep 2012 10:24:22 -0700
curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> You can use modules like mod_enum or mod_easyroute to route the calls
> first before hitting your SIP provider. The wiki is a great place to
> start for this type of scenario

Thanks.  Yes, lots of stuff in the wiki.  The problem is finding what
you need in it.  Pointers like this are very helpful.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From ssinyagin at yahoo.com  Mon Sep 24 00:27:51 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sun, 23 Sep 2012 13:27:51 -0700 (PDT)
Subject: [Freeswitch-users] Some questions from a relative newbie
In-Reply-To: <20120923160739.ee148fb9f6f861bc931dde19@Vex.Net>
References: <20120923103750.543c14ff1599ab5fbfc0427e@Vex.Net>
	<1348424699.12793.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<20120923160739.ee148fb9f6f861bc931dde19@Vex.Net>
Message-ID: <1348432071.18996.YahooMailNeo@web39301.mail.mud.yahoo.com>



in this case, you would have two OpenSIPS boxes and two FreeSwitch boxes. The easiest scenario would be to run them in 
active/standby mode, so that only one box is handling the calls, and the other one starts acting only when it realizes that the master box is 
gone.
Here is one of possible examples:
http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS

There are also some scenarios for active/active redundancy, but that requires much more engineering efforts.




>________________________________
> From: D'Arcy Cain <darcy at Vex.Net>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Cc: Stanislav Sinyagin <ssinyagin at yahoo.com> 
>Sent: Sunday, September 23, 2012 10:07 PM
>Subject: Re: [Freeswitch-users] Some questions from a relative newbie
> 
>On Sun, 23 Sep 2012 11:24:59 -0700 (PDT)
>Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>
>> by the way, do you consider running a fault-tolerant service with
>> OpenSIPS/Kamailio/Repro as a border controller and two FreeSWITCH boxes
>> behind?
>
>Thanks.? I have added that to my reading list.? However, I am not sure
>how useful that scenario would be.? Doesn't the OpenSIPS router just
>become my new point of failure?? If I add another box I can get my
>provider to fail over to it.? That makes them the point of failure but
>they already are so I kind of depend on them not to go down anyway.
>
>-- 
>D'Arcy J.M. Cain
>System Administrator, Vex.Net
>http://www.Vex.Net/ IM:darcy at Vex.Net
>
>
>
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From darcy at Vex.Net  Mon Sep 24 00:04:41 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Sun, 23 Sep 2012 16:04:41 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
Message-ID: <20120923160441.fe473e06925857e1efa7605d@Vex.Net>

On Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> here's a piece of my dialplan in "default" context. It simply catches
> such calls and transfers them directly to the public context. The
> public context matches local DID numbers and transfers them to
> corresponding extensions in default context:
> 
> ??? <extension name="internal_did">
> ????? <condition field="destination_number" expression="^($${default_did_prefix}\d+)$">
> ??????? <action application="transfer" data="$1 XML public"/>
> ????? </condition>
> ??? </extension>
> 
> The "default_did_prefix" variable holds my DID number block. 

Looks good but my problem is that I do not have a DID block.  Clients
are porting their numbers so they are all over the place.  My first
three numbers in the system are from three different area codes.  I
would need to include all of my numbers in the expression somehow.  Wit
just three I can use '|' but that won't be practical when I have
hundreds or thousands.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From darcy at Vex.Net  Mon Sep 24 00:07:39 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Sun, 23 Sep 2012 16:07:39 -0400
Subject: [Freeswitch-users] Some questions from a relative newbie
In-Reply-To: <1348424699.12793.YahooMailNeo@web39304.mail.mud.yahoo.com>
References: <20120923103750.543c14ff1599ab5fbfc0427e@Vex.Net>
	<1348424699.12793.YahooMailNeo@web39304.mail.mud.yahoo.com>
Message-ID: <20120923160739.ee148fb9f6f861bc931dde19@Vex.Net>

On Sun, 23 Sep 2012 11:24:59 -0700 (PDT)
Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> by the way, do you consider running a fault-tolerant service with
> OpenSIPS/Kamailio/Repro as a border controller and two FreeSWITCH boxes
> behind?

Thanks.  I have added that to my reading list.  However, I am not sure
how useful that scenario would be.  Doesn't the OpenSIPS router just
become my new point of failure?  If I add another box I can get my
provider to fail over to it.  That makes them the point of failure but
they already are so I kind of depend on them not to go down anyway.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From ssinyagin at yahoo.com  Mon Sep 24 00:31:29 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sun, 23 Sep 2012 13:31:29 -0700 (PDT)
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <20120923160441.fe473e06925857e1efa7605d@Vex.Net>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
Message-ID: <1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>

well, that was only a hint, but not the whole solution :)
In your case, it would require some database lookup, or maybe the dialplan should go through the customer numbers before sending the call to PSTN.?

enum would be a good option instead of database lookup, as was already proposed on the list.

In any case, this all requires a bit of engineering and design effort. 






>________________________________
> From: D'Arcy Cain <darcy at Vex.Net>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Cc: Stanislav Sinyagin <ssinyagin at yahoo.com> 
>Sent: Sunday, September 23, 2012 10:04 PM
>Subject: Re: [Freeswitch-users] Keeping local calls local
> 
>On Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
>Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>
>> here's a piece of my dialplan in "default" context. It simply catches
>> such calls and transfers them directly to the public context. The
>> public context matches local DID numbers and transfers them to
>> corresponding extensions in default context:
>> 
>> ??? <extension name="internal_did">
>> ????? <condition field="destination_number" expression="^($${default_did_prefix}\d+)$">
>> ??????? <action application="transfer" data="$1 XML public"/>
>> ????? </condition>
>> ??? </extension>
>> 
>> The "default_did_prefix" variable holds my DID number block. 
>
>Looks good but my problem is that I do not have a DID block.? Clients
>are porting their numbers so they are all over the place.? My first
>three numbers in the system are from three different area codes.? I
>would need to include all of my numbers in the expression somehow.? Wit
>just three I can use '|' but that won't be practical when I have
>hundreds or thousands.
>
>-- 
>D'Arcy J.M. Cain
>System Administrator, Vex.Net
>http://www.Vex.Net/ IM:darcy at Vex.Net
>
>
>
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From mario_fs at mgtech.com  Mon Sep 24 01:06:07 2012
From: mario_fs at mgtech.com (Mario G)
Date: Sun, 23 Sep 2012 14:06:07 -0700
Subject: [Freeswitch-users] FreeSwitch Video Calls Bria iPhone Client
In-Reply-To: <2B4CD419-FE1B-443C-86AB-2E5A98C63ED2@me.com>
References: <2B4CD419-FE1B-443C-86AB-2E5A98C63ED2@me.com>
Message-ID: <603F099C-8BDB-4EB5-9375-6DEF87CF06EE@mgtech.com>

FWIW, I was trying to replace SPA962s with Bria on iPads in our office. Had several issues, at least 6 open problems with Counterpath, they were great at trying to help but pointed the finger at FreeSwitch. I posted here for months to no avail so I assume no one here using Bria to this extent. FYI, all my issues are ringing/answering related (listed below). If you find anything please post here as I plan t do the same to keep someone else from pulling their hair out.

1. Bria does not always ring the iPad so you can't answer. I am using TCP to avoid battery drain although UDP did not seem to help. 
2. Bria rings, but when you hit answer there is nothing there and the other extensions keep ringing. FS trace shows FS disconnecting.
3. If Bria is not in the foreground and a ringing alert is displayed, pressing it sometimes results in the last issue of not answering (FS says hangup) but other phones ring.
4. Not a Bria or FS issue but a pain: Apparently Apple limits the alert to 25 seconds, if you don't press it in that time it closed and you can't answer. This is an IOS issue that Apple needs to allow more flexibility with.

Once a call is connected there are no issues with the recent Bria iPad updates, there were some issues a month ago. I highly recommend opening an issue with Counterpath as they seemed to know SIP and understood some of the FS trace stuff. They are very quick to update the Bria if an issue is found. But in my case, after 6+ months of traces we can't find the culprit.

I have no Nat issues at all using Zyxel routers with SIP support. Good luck to you, I am writing the president of CounterPath to see if they can get a FreeSwitch to test with since it would be great for both to work together.
Mario G

On Sep 22, 2012, at 7:46 PM, Mike Burlingame wrote:

> I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android? 
> 
> 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 lconroy at insensate.co.uk  Mon Sep 24 01:54:53 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Sun, 23 Sep 2012 22:54:53 +0100
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <20120923160441.fe473e06925857e1efa7605d@Vex.Net>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
Message-ID: <C067307A-A078-44DE-A922-37F436C3AA36@insensate.co.uk>

Hi there,
 full disclosure -- on ENUM, I'm biased :).
Seriously, if you think you'll have a lot of users, and those users will be spread over the map, a relatively straightforward way of doing this is to have your own private ENUM tree in DNS (apex of which could be in e164.vex.net, for example).

You provision the sub-domains matching your user's numbers with E2U+sip NAPTRs pointing at your server.
If there IS no hit in this private ENUM tree, fire the call off to your provider.
Your users dial the "full" number, that full number hits the ENUM tree, and if the DN is one of yours, mod_enum will extract the SIP URL from the matching DNS domain and does its stuff. Done.
(you can do this with a very basic XML dialplan -- just make sure that the ENUM lookup comes ->first<-, and that the dialplan processing continues if there is no ENUM hit).

Cost -- a local DNS server plus some script to provision the DNS record into your private DNS tree.
[and a mental squint headache when writing the script to converting from a phone number like +15552345987 to 7.8.9.5.4.3.2.5.5.5.1.e164.vex.net.]
I've used BIND/DLZ&PSQL, but Bert's PowerDNS should be fine as well.

Remember that you'll be doing a DNS query for every call attempt, so DO keep the DNS server "close", and don't use some "less than perfect" DNS software.

Is it worth it for a few 10s of users? Nah (unless you're a propellor head, or expect to grow).
Is it worth it for hundreds of users? Yup.
Is it worth it for thousands or millions of users? [I have reason to believe this is what a certain nationwide cable company C does internally for its customers across the 'States -- avoiding number dips saves a shedload of money]

Just a thought.

all the best,
  Lawrence
[author of RFC6116 ;]

On 23 Sep 2012, at 21:04, D'Arcy Cain wrote:
> On Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
> Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> 
>> here's a piece of my dialplan in "default" context. It simply catches
>> such calls and transfers them directly to the public context. The
>> public context matches local DID numbers and transfers them to
>> corresponding extensions in default context:
>> 
>>     <extension name="internal_did">
>>       <condition field="destination_number" expression="^($${default_did_prefix}\d+)$">
>>         <action application="transfer" data="$1 XML public"/>
>>       </condition>
>>     </extension>
>> 
>> The "default_did_prefix" variable holds my DID number block. 
> 
> Looks good but my problem is that I do not have a DID block.  Clients
> are porting their numbers so they are all over the place.  My first
> three numbers in the system are from three different area codes.  I
> would need to include all of my numbers in the expression somehow.  Wit
> just three I can use '|' but that won't be practical when I have
> hundreds or thousands.
> 
> -- 
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.Net



From ssinyagin at yahoo.com  Mon Sep 24 02:00:04 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sun, 23 Sep 2012 15:00:04 -0700 (PDT)
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
Message-ID: <1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>

Basically your inbouind and outbound calls should go through the same sequence of checkups, and look for a local user number. Then outbound calls should resort in your PSTN trunk, and inbound calls resort in a 404 Not Found. Also I guess you want to charge your users the same rate regardless if it's on-net or off-net calls, so CDR need to be taken care of too.





>________________________________
> From: Stanislav Sinyagin <ssinyagin at yahoo.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Sunday, September 23, 2012 10:31 PM
>Subject: Re: [Freeswitch-users] Keeping local calls local
> 
>
>well, that was only a hint, but not the whole solution :)
>In your case, it would require some database lookup, or maybe the dialplan should go through the customer numbers before sending the call to PSTN.?
>
>
>enum would be a good option instead of database lookup, as was already proposed on the list.
>
>
>In any case, this all requires a bit of engineering and design effort. 
>
>
>
>
>
>
>
>
>>________________________________
>> From: D'Arcy Cain <darcy at Vex.Net>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>Cc: Stanislav Sinyagin <ssinyagin at yahoo.com> 
>>Sent: Sunday, September 23, 2012 10:04 PM
>>Subject: Re: [Freeswitch-users] Keeping local calls local
>> 
>>On Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
>>Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>
>>> here's a piece of my dialplan in "default" context. It simply catches
>>> such calls and transfers them directly to the public context. The
>>> public context matches local DID numbers and transfers them to
>>> corresponding extensions in default context:
>>> 
>>> ??? <extension name="internal_did">
>>> ????? <condition field="destination_number" expression="^($${default_did_prefix}\d+)$">
>>> ??????? <action application="transfer" data="$1 XML public"/>
>>> ????? </condition>
>>> ??? </extension>
>>> 
>>> The "default_did_prefix" variable holds my DID number block. 
>>
>>Looks good
 but my problem is that I do not have a DID block.? Clients
>>are porting their numbers so they are all over the place.? My first
>>three numbers in the system are from three different area codes.? I
>>would need to include all of my numbers in the expression somehow.? Wit
>>just three I can use '|' but that won't be practical when I have
>>hundreds or thousands.
>>
>>-- 
>>D'Arcy J.M. Cain
>>System Administrator, Vex.Net
>>http://www.Vex.Net/ IM:darcy at Vex.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
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.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 Sep 24 02:03:24 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 23 Sep 2012 18:03:24 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
Message-ID: <CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>

Mod_lcr?
On Sep 23, 2012 6:01 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:

> Basically your inbouind and outbound calls should go through the same
> sequence of checkups, and look for a local user number. Then outbound calls
> should resort in your PSTN trunk, and inbound calls resort in a 404 Not
> Found. Also I guess you want to charge your users the same rate regardless
> if it's on-net or off-net calls, so CDR need to be taken care of too.
>
>
>   ------------------------------
> *From:* Stanislav Sinyagin <ssinyagin at yahoo.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, September 23, 2012 10:31 PM
> *Subject:* Re: [Freeswitch-users] Keeping local calls local
>
> well, that was only a hint, but not the whole solution :)
> In your case, it would require some database lookup, or maybe the dialplan
> should go through the customer numbers before sending the call to PSTN.
>
> enum would be a good option instead of database lookup, as was already
> proposed on the list.
>
> In any case, this all requires a bit of engineering and design effort.
>
>
>
>   ------------------------------
> *From:* D'Arcy Cain <darcy at Vex.Net>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Cc:* Stanislav Sinyagin <ssinyagin at yahoo.com>
> *Sent:* Sunday, September 23, 2012 10:04 PM
> *Subject:* Re: [Freeswitch-users] Keeping local calls local
>
> On Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
> Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>
> > here's a piece of my dialplan in "default" context. It simply catches
> > such calls and transfers them directly to the public context. The
> > public context matches local DID numbers and transfers them to
> > corresponding extensions in default context:
> >
> >     <extension name="internal_did">
> >       <condition field="destination_number"
> expression="^($${default_did_prefix}\d+)$">
> >         <action application="transfer" data="$1 XML public"/>
> >       </condition>
> >     </extension>
> >
> > The "default_did_prefix" variable holds my DID number block.
>
> Looks good but my problem is that I do not have a DID block.  Clients
> are porting their numbers so they are all over the place.  My first
> three numbers in the system are from three different area codes.  I
> would need to include all of my numbers in the expression somehow.  Wit
> just three I can use '|' but that won't be practical when I have
> hundreds or thousands.
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Sep 24 02:35:41 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sun, 23 Sep 2012 16:35:41 -0600
Subject: [Freeswitch-users] Building v1.2 .debs on Ubuntu 12.04 LTS
In-Reply-To: <505EC595.8010700@quentustech.com>
References: <CAMwZ-t+bqZ6NUc5f7z2hFB3dFcNpT=86v3kfH_62Ws=M_Rs6yg@mail.gmail.com>
	<505EC595.8010700@quentustech.com>
Message-ID: <CAMwZ-tKgXy4CHSELDND8zJj3heJ0f_yNhngktGSB7Nst=h2d4g@mail.gmail.com>

On Sun, Sep 23, 2012 at 2:17 AM, William King
<william.king at quentustech.com> wrote:
> There are scripts for building the dsc and deb files in tree right now.
> check the ./scripts/ci/ folder.

Very nice, I hadn't seen those. I'll give a them a spin and report :)


Gabe


From lconroy at insensate.co.uk  Mon Sep 24 02:36:00 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Sun, 23 Sep 2012 23:36:00 +0100
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
Message-ID: <1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>

Hi folks,
 Fair enough, but ... the original problem is that the users are/will be porting their current phone numbers, so they're all over the map.
That's going to make routing/pattern matching a challenge, because there IS no obvious routeset pattern. Honestly, ENUM is easier in that scenario.

all the best,
  Lawrence

On 23 Sep 2012, at 23:03, Brian Foster wrote:
> Mod_lcr?
> On Sep 23, 2012 6:01 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
> 
>> Basically your inbouind and outbound calls should go through the same
>> sequence of checkups, and look for a local user number. Then outbound calls
>> should resort in your PSTN trunk, and inbound calls resort in a 404 Not
>> Found. Also I guess you want to charge your users the same rate regardless
>> if it's on-net or off-net calls, so CDR need to be taken care of too.
>> 
>> 
>>  ------------------------------
>> *From:* Stanislav Sinyagin <ssinyagin at yahoo.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Sunday, September 23, 2012 10:31 PM
>> *Subject:* Re: [Freeswitch-users] Keeping local calls local
>> 
>> well, that was only a hint, but not the whole solution :)
>> In your case, it would require some database lookup, or maybe the dialplan
>> should go through the customer numbers before sending the call to PSTN.
>> 
>> enum would be a good option instead of database lookup, as was already
>> proposed on the list.
>> 
>> In any case, this all requires a bit of engineering and design effort.
>> 
>> 
>> 
>>  ------------------------------
>> *From:* D'Arcy Cain <darcy at Vex.Net>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Cc:* Stanislav Sinyagin <ssinyagin at yahoo.com>
>> *Sent:* Sunday, September 23, 2012 10:04 PM
>> *Subject:* Re: [Freeswitch-users] Keeping local calls local
>> 
>> On Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
>> Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>> 
>>> here's a piece of my dialplan in "default" context. It simply catches
>>> such calls and transfers them directly to the public context. The
>>> public context matches local DID numbers and transfers them to
>>> corresponding extensions in default context:
>>> 
>>>    <extension name="internal_did">
>>>      <condition field="destination_number"
>> expression="^($${default_did_prefix}\d+)$">
>>>        <action application="transfer" data="$1 XML public"/>
>>>      </condition>
>>>    </extension>
>>> 
>>> The "default_did_prefix" variable holds my DID number block.
>> 
>> Looks good but my problem is that I do not have a DID block.  Clients
>> are porting their numbers so they are all over the place.  My first
>> three numbers in the system are from three different area codes.  I
>> would need to include all of my numbers in the expression somehow.  Wit
>> just three I can use '|' but that won't be practical when I have
>> hundreds or thousands.
>> 
>> --
>> D'Arcy J.M. Cain
>> System Administrator, Vex.Net
>> http://www.Vex.Net/ IM:darcy at Vex.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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon Sep 24 02:40:37 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 24 Sep 2012 00:40:37 +0200
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
	<1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>
Message-ID: <CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>

enum or a DB query to your routing table for inbound calls, e.g. using
mod_odbc_query <http://wiki.freeswitch.org/wiki/Mod_odbc_query>.

-Avi

On Mon, Sep 24, 2012 at 12:36 AM, Lawrence Conroy
<lconroy at insensate.co.uk>wrote:

> Hi folks,
>  Fair enough, but ... the original problem is that the users are/will be
> porting their current phone numbers, so they're all over the map.
> That's going to make routing/pattern matching a challenge, because there
> IS no obvious routeset pattern. Honestly, ENUM is easier in that scenario.
>
> all the best,
>   Lawrence
>
> On 23 Sep 2012, at 23:03, Brian Foster wrote:
> > Mod_lcr?
> > On Sep 23, 2012 6:01 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com>
> wrote:
> >
> >> Basically your inbouind and outbound calls should go through the same
> >> sequence of checkups, and look for a local user number. Then outbound
> calls
> >> should resort in your PSTN trunk, and inbound calls resort in a 404 Not
> >> Found. Also I guess you want to charge your users the same rate
> regardless
> >> if it's on-net or off-net calls, so CDR need to be taken care of too.
> >>
> >>
> >>  ------------------------------
> >> *From:* Stanislav Sinyagin <ssinyagin at yahoo.com>
> >> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> *Sent:* Sunday, September 23, 2012 10:31 PM
> >> *Subject:* Re: [Freeswitch-users] Keeping local calls local
> >>
> >> well, that was only a hint, but not the whole solution :)
> >> In your case, it would require some database lookup, or maybe the
> dialplan
> >> should go through the customer numbers before sending the call to PSTN.
> >>
> >> enum would be a good option instead of database lookup, as was already
> >> proposed on the list.
> >>
> >> In any case, this all requires a bit of engineering and design effort.
> >>
> >>
> >>
> >>  ------------------------------
> >> *From:* D'Arcy Cain <darcy at Vex.Net>
> >> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> *Cc:* Stanislav Sinyagin <ssinyagin at yahoo.com>
> >> *Sent:* Sunday, September 23, 2012 10:04 PM
> >> *Subject:* Re: [Freeswitch-users] Keeping local calls local
> >>
> >> On Sun, 23 Sep 2012 11:17:37 -0700 (PDT)
> >> Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> >>
> >>> here's a piece of my dialplan in "default" context. It simply catches
> >>> such calls and transfers them directly to the public context. The
> >>> public context matches local DID numbers and transfers them to
> >>> corresponding extensions in default context:
> >>>
> >>>    <extension name="internal_did">
> >>>      <condition field="destination_number"
> >> expression="^($${default_did_prefix}\d+)$">
> >>>        <action application="transfer" data="$1 XML public"/>
> >>>      </condition>
> >>>    </extension>
> >>>
> >>> The "default_did_prefix" variable holds my DID number block.
> >>
> >> Looks good but my problem is that I do not have a DID block.  Clients
> >> are porting their numbers so they are all over the place.  My first
> >> three numbers in the system are from three different area codes.  I
> >> would need to include all of my numbers in the expression somehow.  Wit
> >> just three I can use '|' but that won't be practical when I have
> >> hundreds or thousands.
> >>
> >> --
> >> D'Arcy J.M. Cain
> >> System Administrator, Vex.Net
> >> http://www.Vex.Net/ IM:darcy at Vex.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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Sep 24 03:07:13 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sun, 23 Sep 2012 17:07:13 -0600
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <C43E3862-427B-454D-AA1C-F59AB3F0F391@gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
	<C43E3862-427B-454D-AA1C-F59AB3F0F391@gmail.com>
Message-ID: <CAMwZ-tLJjf02vACgORBebergpQssftUOU=WT=69dc_DeiunOJQ@mail.gmail.com>

On Sun, Sep 23, 2012 at 9:17 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
> The part although not the same website is on;
>
> https://profiles.google.com/100965631603901259030/buzz
>
> "Another tip is to use cacheable='true' attribute of user xml returned by
> web service. This will make FS to maintain a local cache of the account
> details. The details will stay in cache until cleared. This means that FS
> won't be hitting your web app for every REGISTER request that hoards of SIP
> UAs are sending, it will ask only once upon first request. The subsequent
> requests will be looked up from cache."

Ok, I see what you're talking about. I think what you're looking at
isn't mod_xml_curl wide. It seems to be specific to user registration.
I'm not sure about this, but I don't see the word 'cacheable' in the
mod_xml_curl source, but you do see it on line 1946 of switch_xml.c.
And there too, it appears to have something to do with users.

If any of that is correct, is suggests that you will not be able to
add "cacheable" to your dialplan served up by mod_xml_curl and
"prevent a post every time a dial is made."

Hope that helps.


Best,
Gabe


From ssinyagin at yahoo.com  Mon Sep 24 03:40:15 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sun, 23 Sep 2012 16:40:15 -0700 (PDT)
Subject: [Freeswitch-users] Keeping local calls local
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net> 
Message-ID: <1348443615.85843.YahooMailNeo@web39305.mail.mud.yahoo.com>

going back to your original message,

1. having a separate condition for every user means that the switch will have to search through hundreds of regular expressions on every call. Not nice and a waste of cpu time. So, you need to look up some database instead.

2. after an unsuccessful bridge, you resort in voicemail. Are you sure all your users want voicemail? Some will prefer the call to be unanswered, and some will want to forward the call if unsuccessful. Also some will want to always forward the calls. So, again you need to look up in user preferences database.

also probably it makes sense to offload the user authentication and location database to an OpenSIPS or some other proxy, and let FreeSWITCH deal with call routing, media and voicemail.




>________________________________
> From: D'Arcy Cain <darcy at Vex.Net>
>To: FreeSWITCH-users at lists.freeswitch.org
>Sent: Sunday, September 23, 2012 5:01 PM
>Subject: [Freeswitch-users] Keeping local calls local
> 
>The first thing I found with the basic setup of FreeSWITCH was that
>calls between my own clients still went to my provider.? This seemed
>like a waste to me.? To handle calls coming in I generate an XML file
>called dialplan/public/00_Local.xml.? Here is an example entry from it.
>
><extension name="Local_4164251212">
>? <condition field="destination_number" expression="^(4164251212)$">
>? ? <action application="export" data="dialed_extension=$1"/>
>? ? <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
>? ? <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>? ? <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
>? ? <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
>? ? <action application="set" data="ringback=${us-ring}"/>
>? ? <action application="set" data="transfer_ringback=$${hold_music}"/>
>? ? <action application="set" data="call_timeout=30"/>
>? ? <action application="set" data="hangup_after_bridge=true"/>
>? ? <action application="set" data="continue_on_fail=true"/>
>? ? <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>? ? <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>? ? <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>? ? <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
>? ? <action application="answer"/>
>? ? <action application="sleep" data="1000"/>
>? ? <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>? </condition>
></extension>
>
>I am still trying to figure out exactly what everything in there does
>but it seems to do the job.? Calls originating outside get delivered to
>the correct client's phone.? Calls between clients go out to the
>provider and then come back in as if it was external.
>
>I worked around this by symlinking this file to the dialplan/default
>directory.? This works but I wonder if this is the best way to
>accomplish this.
>
>Side question, can I create a simple alias for all of my DIDs and make
>one condition that tests that alias somehow?
>
>Cheers.
>
>-- 
>D'Arcy J.M. Cain
>System Administrator, Vex.Net
>http://www.Vex.Net/ IM:darcy at Vex.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
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>



From jaybinks at gmail.com  Mon Sep 24 04:39:43 2012
From: jaybinks at gmail.com (jay binks)
Date: Mon, 24 Sep 2012 10:39:43 +1000
Subject: [Freeswitch-users] FreeSwitch Video Calls Bria iPhone Client
In-Reply-To: <603F099C-8BDB-4EB5-9375-6DEF87CF06EE@mgtech.com>
References: <2B4CD419-FE1B-443C-86AB-2E5A98C63ED2@me.com>
	<603F099C-8BDB-4EB5-9375-6DEF87CF06EE@mgtech.com>
Message-ID: <CAO47HpeOng3U4TE8TPLY6JcAyTvJYiufYpfUT30qz3OgNqhLNA@mail.gmail.com>

You know your zyxel  could be the issue  for lots of that .  Sip aware Nat
router = sip ALG = black magic going on that neither end can predict.

Remove or disable the alg  and try again .
On Sep 24, 2012 4:10 AM, "Mario G" <mario_fs at mgtech.com> wrote:

> FWIW, I was trying to replace SPA962s with Bria on iPads in our office.
> Had several issues, at least 6 open problems with Counterpath, they were
> great at trying to help but pointed the finger at FreeSwitch. I posted here
> for months to no avail so I assume no one here using Bria to this extent.
> FYI, all my issues are ringing/answering related (listed below). If you
> find anything please post here as I plan t do the same to keep someone else
> from pulling their hair out.
>
> 1. Bria does not always ring the iPad so you can't answer. I am using TCP
> to avoid battery drain although UDP did not seem to help.
> 2. Bria rings, but when you hit answer there is nothing there and the
> other extensions keep ringing. FS trace shows FS disconnecting.
> 3. If Bria is not in the foreground and a ringing alert is displayed,
> pressing it sometimes results in the last issue of not answering (FS says
> hangup) but other phones ring.
> 4. Not a Bria or FS issue but a pain: Apparently Apple limits the alert to
> 25 seconds, if you don't press it in that time it closed and you can't
> answer. This is an IOS issue that Apple needs to allow more flexibility
> with.
>
> Once a call is connected there are no issues with the recent Bria iPad
> updates, there were some issues a month ago. I highly recommend opening an
> issue with Counterpath as they seemed to know SIP and understood some of
> the FS trace stuff. They are very quick to update the Bria if an issue is
> found. But in my case, after 6+ months of traces we can't find the culprit.
>
> I have no Nat issues at all using Zyxel routers with SIP support. Good
> luck to you, I am writing the president of CounterPath to see if they can
> get a FreeSwitch to test with since it would be great for both to work
> together.
> Mario G
>
> On Sep 22, 2012, at 7:46 PM, Mike Burlingame wrote:
>
> > I have been playing around trying to get my iPad and iPhone using Bria
> to work with video it *seems* that FreeSwitch is passing the signaling to
> both devices and FreeSwitch is holding the media due to the NAT issue of
> the devices - I just wanted to know before I spent too much time on this
> (not a high priority) if anyone else has gotten FreeSwitch to work with
> Bria for iPhone or Android?
> >
> > 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 bdfoster at endigotech.com  Mon Sep 24 05:08:32 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 23 Sep 2012 21:08:32 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <1348443615.85843.YahooMailNeo@web39305.mail.mud.yahoo.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348443615.85843.YahooMailNeo@web39305.mail.mud.yahoo.com>
Message-ID: <CAKV-jFxau-Ey60-j-Y58ugyGP1PktQVWXTVD1HS+q+unoiO1sw@mail.gmail.com>

Im still liking the mod_lcr idea since you should be doing it anyway for
something like this.
On Sep 23, 2012 7:41 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:

> going back to your original message,
>
> 1. having a separate condition for every user means that the switch will
> have to search through hundreds of regular expressions on every call. Not
> nice and a waste of cpu time. So, you need to look up some database instead.
>
> 2. after an unsuccessful bridge, you resort in voicemail. Are you sure all
> your users want voicemail? Some will prefer the call to be unanswered, and
> some will want to forward the call if unsuccessful. Also some will want to
> always forward the calls. So, again you need to look up in user preferences
> database.
>
> also probably it makes sense to offload the user authentication and
> location database to an OpenSIPS or some other proxy, and let FreeSWITCH
> deal with call routing, media and voicemail.
>
>
>
>
> >________________________________
> > From: D'Arcy Cain <darcy at Vex.Net>
> >To: FreeSWITCH-users at lists.freeswitch.org
> >Sent: Sunday, September 23, 2012 5:01 PM
> >Subject: [Freeswitch-users] Keeping local calls local
> >
> >The first thing I found with the basic setup of FreeSWITCH was that
> >calls between my own clients still went to my provider.  This seemed
> >like a waste to me.  To handle calls coming in I generate an XML file
> >called dialplan/public/00_Local.xml.  Here is an example entry from it.
> >
> ><extension name="Local_4164251212">
> >  <condition field="destination_number" expression="^(4164251212)$">
> >    <action application="export" data="dialed_extension=$1"/>
> >    <action application="bind_meta_app" data="1 b s execute_extension::dx
> XML features"/>
> >    <action application="bind_meta_app" data="2 b s
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> >    <action application="bind_meta_app" data="3 b s execute_extension::cf
> XML features"/>
> >    <action application="bind_meta_app" data="4 b s
> execute_extension::att_xfer XML features"/>
> >    <action application="set" data="ringback=${us-ring}"/>
> >    <action application="set" data="transfer_ringback=$${hold_music}"/>
> >    <action application="set" data="call_timeout=30"/>
> >    <action application="set" data="hangup_after_bridge=true"/>
> >    <action application="set" data="continue_on_fail=true"/>
> >    <action application="hash"
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
> >    <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
> >    <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)}"/>
> >    <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
> >    <action application="hash"
> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
> >    <action application="hash"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
> >    <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
> >    <action application="answer"/>
> >    <action application="sleep" data="1000"/>
> >    <action application="bridge" data="loopback/app=voicemail:default
> ${domain_name} ${dialed_extension}"/>
> >  </condition>
> ></extension>
> >
> >I am still trying to figure out exactly what everything in there does
> >but it seems to do the job.  Calls originating outside get delivered to
> >the correct client's phone.  Calls between clients go out to the
> >provider and then come back in as if it was external.
> >
> >I worked around this by symlinking this file to the dialplan/default
> >directory.  This works but I wonder if this is the best way to
> >accomplish this.
> >
> >Side question, can I create a simple alias for all of my DIDs and make
> >one condition that tests that alias somehow?
> >
> >Cheers.
> >
> >--
> >D'Arcy J.M. Cain
> >System Administrator, Vex.Net
> >http://www.Vex.Net/ IM:darcy at Vex.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
> >
> >FreeSWITCH-users mailing list
> >FreeSWITCH-users at lists.freeswitch.org
> >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >http://www.freeswitch.org
> >
> >
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From zhulizhong at live.com  Mon Sep 24 05:48:22 2012
From: zhulizhong at live.com (James zhu)
Date: Mon, 24 Sep 2012 01:48:22 +0000
Subject: [Freeswitch-users] Freeswitch+sangoma a101 can not hear ring back
	tone
Message-ID: <BLU171-W1646AE3824D87F824C0990B59E0@phx.gbl>


hello:I install fw and sangoma A101 card with sangoma isdn. after few days, caller can not hear the ring back tone, but the voice worked. the hardware is ok, compare with normal status, the alert message lost. how do i set that to enforce to send alert message?

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway.
website: www.hiastar.com 
 		 	   		  
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From mike.burlingame at me.com  Mon Sep 24 05:59:42 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sun, 23 Sep 2012 18:59:42 -0700
Subject: [Freeswitch-users] FreeSwitch Video Calls Bria iPhone Client
In-Reply-To: <CAO47HpeOng3U4TE8TPLY6JcAyTvJYiufYpfUT30qz3OgNqhLNA@mail.gmail.com>
References: <2B4CD419-FE1B-443C-86AB-2E5A98C63ED2@me.com>
	<603F099C-8BDB-4EB5-9375-6DEF87CF06EE@mgtech.com>
	<CAO47HpeOng3U4TE8TPLY6JcAyTvJYiufYpfUT30qz3OgNqhLNA@mail.gmail.com>
Message-ID: <7DE0AD3C-2A16-49B6-8533-3F0834D54768@me.com>

I agree I am not seeing any issues on call setup / take down or audio RTP or SMS only issues with the video setting up correctly

Have you tried to remove the ALG and reproduced the same issue?

Sent from my iPhone 4S

On Sep 23, 2012, at 5:39 PM, jay binks <jaybinks at gmail.com> wrote:

> You know your zyxel  could be the issue  for lots of that .  Sip aware Nat router = sip ALG = black magic going on that neither end can predict.
> 
> Remove or disable the alg  and try again .
> 
> On Sep 24, 2012 4:10 AM, "Mario G" <mario_fs at mgtech.com> wrote:
>> FWIW, I was trying to replace SPA962s with Bria on iPads in our office. Had several issues, at least 6 open problems with Counterpath, they were great at trying to help but pointed the finger at FreeSwitch. I posted here for months to no avail so I assume no one here using Bria to this extent. FYI, all my issues are ringing/answering related (listed below). If you find anything please post here as I plan t do the same to keep someone else from pulling their hair out.
>> 
>> 1. Bria does not always ring the iPad so you can't answer. I am using TCP to avoid battery drain although UDP did not seem to help.
>> 2. Bria rings, but when you hit answer there is nothing there and the other extensions keep ringing. FS trace shows FS disconnecting.
>> 3. If Bria is not in the foreground and a ringing alert is displayed, pressing it sometimes results in the last issue of not answering (FS says hangup) but other phones ring.
>> 4. Not a Bria or FS issue but a pain: Apparently Apple limits the alert to 25 seconds, if you don't press it in that time it closed and you can't answer. This is an IOS issue that Apple needs to allow more flexibility with.
>> 
>> Once a call is connected there are no issues with the recent Bria iPad updates, there were some issues a month ago. I highly recommend opening an issue with Counterpath as they seemed to know SIP and understood some of the FS trace stuff. They are very quick to update the Bria if an issue is found. But in my case, after 6+ months of traces we can't find the culprit.
>> 
>> I have no Nat issues at all using Zyxel routers with SIP support. Good luck to you, I am writing the president of CounterPath to see if they can get a FreeSwitch to test with since it would be great for both to work together.
>> Mario G
>> 
>> On Sep 22, 2012, at 7:46 PM, Mike Burlingame wrote:
>> 
>> > I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android?
>> >
>> > 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 anthony.minessale at gmail.com  Mon Sep 24 07:19:53 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 23 Sep 2012 22:19:53 -0500
Subject: [Freeswitch-users] Some questions from a relative newbie
In-Reply-To: <1348432071.18996.YahooMailNeo@web39301.mail.mud.yahoo.com>
References: <20120923103750.543c14ff1599ab5fbfc0427e@Vex.Net>
	<1348424699.12793.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<20120923160739.ee148fb9f6f861bc931dde19@Vex.Net>
	<1348432071.18996.YahooMailNeo@web39301.mail.mud.yahoo.com>
Message-ID: <CAKbxfG-7G5fmh9wbHpLr37ojYAsog_rvMiPY2o0w=Qz1UiRWLg@mail.gmail.com>

If you start it with -ncwait it will block until its 100% up then fork into
the bg.
I don't think fs_cli ever behaved any differently but before this option
was added about a year ago, and unless its specified, just -nc will return
right away before its even close to ready.


On Sun, Sep 23, 2012 at 3:27 PM, Stanislav Sinyagin <ssinyagin at yahoo.com>wrote:

> in this case, you would have two OpenSIPS boxes and two FreeSwitch boxes.
> The easiest scenario would be to run them in active/standby mode, so that
> only one box is handling the calls, and the other one starts acting only
> when it realizes that the master box is gone.
> Here is one of possible examples:
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS
>
> There are also some scenarios for active/active redundancy, but that
> requires much more engineering efforts.
>
>
>   ------------------------------
> *From:* D'Arcy Cain <darcy at Vex.Net>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Cc:* Stanislav Sinyagin <ssinyagin at yahoo.com>
> *Sent:* Sunday, September 23, 2012 10:07 PM
> *Subject:* Re: [Freeswitch-users] Some questions from a relative newbie
>
> On Sun, 23 Sep 2012 11:24:59 -0700 (PDT)
> Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>
> > by the way, do you consider running a fault-tolerant service with
> > OpenSIPS/Kamailio/Repro as a border controller and two FreeSWITCH boxes
> > behind?
>
> Thanks.  I have added that to my reading list.  However, I am not sure
> how useful that scenario would be.  Doesn't the OpenSIPS router just
> become my new point of failure?  If I add another box I can get my
> provider to fail over to it.  That makes them the point of failure but
> they already are so I kind of depend on them not to go down anyway.
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Mon Sep 24 07:46:11 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 23 Sep 2012 22:46:11 -0500
Subject: [Freeswitch-users] FreeSWITCH HA Database changes in 1.2
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900146996AD5@EX.frontier.local>
References: <0D1C698866F66045A6201FD0F59CAC900146996AD5@EX.frontier.local>
Message-ID: <CAKbxfG--pJq4VTrO4WPCRM_+iWCEviQo7Vg7Nvt730nF7B+V_w@mail.gmail.com>

On Fri, Sep 21, 2012 at 6:33 PM, Colin Mason <cmason at frontiernetworks.ca>wrote:

> Hello,****
>
> ** **
>
> I noticed recently on FS 1.2 that the table ?sip_recovery? wasn?t being
> created automatically by FreeSWITCH. If I specify:****
>
> ** **
>
> <param name="core-recovery-db-dsn"
> value="maxpowersoft_odbc:freeswitch:blahblah"/>****
>
> ** **
>
> In switch.conf.xml, FreeSWITCH will create a table called ?recovery?.****
>
> ** **
>
> ** **
>
> ** **
>
> So my questions are:****
>
> ** **
>
> **1.       **Have you renamed the sip_recovery table to recovery?****
>
> **
>
Yes its been centralized to the core and abstracted so other endpoints can
use it.





> 2.       **Is ?core-recovery-db-dsn? a required parameter for HA
> FreeSWITCH to function?****
>
> **
>

Its required if you choose to use odbc for your recovery table instead of
sqlite.  Basically set it to the same value as odbc-dsn if that is what you
were using with sofia, or give it a dedicated database.



>  **
>
> Thanks,****
>
> Colin****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 yehavi.bourvine at gmail.com  Mon Sep 24 07:53:29 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 24 Sep 2012 05:53:29 +0200
Subject: [Freeswitch-users] Softphone with callee name display?
Message-ID: <CAAtWKeHznz7bLSMnLJ_tn=BZH=mFPO5NDb9S0q0kWWZ8GrU1nA@mail.gmail.com>

Hello,

  Polycom and Snom (at least) can display the name of the called person
(sent by P-Remote-ID or P-Asserted-ID fields). Is there a softphone that
does the same? I couldn't find suh one so far...

                  Thanks! __Yehavi:
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From mitch.capper at gmail.com  Mon Sep 24 10:26:08 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 24 Sep 2012 08:26:08 +0200
Subject: [Freeswitch-users] Softphone with callee name display?
In-Reply-To: <CAAtWKeHznz7bLSMnLJ_tn=BZH=mFPO5NDb9S0q0kWWZ8GrU1nA@mail.gmail.com>
References: <CAAtWKeHznz7bLSMnLJ_tn=BZH=mFPO5NDb9S0q0kWWZ8GrU1nA@mail.gmail.com>
Message-ID: <CALn233q=w2NvBxm+Hzo5VcOr3h1fwAWPGbkC_gUVq=KMupa+vg@mail.gmail.com>

FSClient shows caller id:)

~Mitch

On Mon, Sep 24, 2012 at 5:53 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Hello,
>
>   Polycom and Snom (at least) can display the name of the called person
> (sent by P-Remote-ID or P-Asserted-ID fields). Is there a softphone that
> does the same? I couldn't find suh one so far...
>
>                   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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peetzer at gmail.com  Mon Sep 24 09:20:21 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Mon, 24 Sep 2012 07:20:21 +0200
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <CAMwZ-tLJjf02vACgORBebergpQssftUOU=WT=69dc_DeiunOJQ@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
	<C43E3862-427B-454D-AA1C-F59AB3F0F391@gmail.com>
	<CAMwZ-tLJjf02vACgORBebergpQssftUOU=WT=69dc_DeiunOJQ@mail.gmail.com>
Message-ID: <CACSMF4zdv0430_YOzdzHwCNhyLQGkWPCijTdn7-suCGM1Y0mXg@mail.gmail.com>

Hi Gabe,

Thanks for looking in to it. I have seen the source as well, with keyword
cacheable.
It is a pity if this can't be used in mod curl.

Regards Peter

On Mon, Sep 24, 2012 at 1:07 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Sep 23, 2012 at 9:17 AM, Peter van Raamsdonk <peetzer at gmail.com>
> wrote:
> > The part although not the same website is on;
> >
> > https://profiles.google.com/100965631603901259030/buzz
> >
> > "Another tip is to use cacheable='true' attribute of user xml returned by
> > web service. This will make FS to maintain a local cache of the account
> > details. The details will stay in cache until cleared. This means that FS
> > won't be hitting your web app for every REGISTER request that hoards of
> SIP
> > UAs are sending, it will ask only once upon first request. The subsequent
> > requests will be looked up from cache."
>
> Ok, I see what you're talking about. I think what you're looking at
> isn't mod_xml_curl wide. It seems to be specific to user registration.
> I'm not sure about this, but I don't see the word 'cacheable' in the
> mod_xml_curl source, but you do see it on line 1946 of switch_xml.c.
> And there too, it appears to have something to do with users.
>
> If any of that is correct, is suggests that you will not be able to
> add "cacheable" to your dialplan served up by mod_xml_curl and
> "prevent a post every time a dial is made."
>
> Hope that helps.
>
>
> 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 peetzer at gmail.com  Mon Sep 24 10:16:42 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Mon, 24 Sep 2012 08:16:42 +0200
Subject: [Freeswitch-users] Fifo api command register agents (add
	fifo_member)
In-Reply-To: <1CD16CE835DC4C7C815E8D8599D81DF3@gmail.com>
References: <9205AA1A-DBDE-4B0F-92B6-37605C4A485E@gmail.com>
	<1CD16CE835DC4C7C815E8D8599D81DF3@gmail.com>
Message-ID: <CACSMF4zCV2MTCnV_KA-_pTysRLy1qF2aGs4Nf_-i4TQaCbF64Q@mail.gmail.com>

Hi Seven,

Api works (esl client) sendSyncApiCommand. When I run fifo list, I see
added member. Sendmsg still does nothing :/

[sendmsg 821701b6-f20d-4721-8414-14e159a39840, call-command: fifo_member,
execute-app-name: add, execute-app-arg:
TestQueue sofia/internal/1001%123.123.123.123 1 60 30]

But better than nothing ;)

Regards Peter

On Sat, Sep 22, 2012 at 1:31 PM, Seven Du <dujinfang at gmail.com> wrote:

> sendmsg is not for api, try
>
>
> api fifo_member add blah????\n\n
>
>
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
>  On Saturday, September 22, 2012 at 3:03 PM, Peter van Raamsdonk wrote:
>
>   Hi all,
>
> I started to implement fifo in my application to handle calls from know
> phones to unknown extensions and specific dial string/ extension.
>
> I use the api commands with esl (java client), outbound socket (with
> sendMsg).
>
> http://wiki.freeswitch.org/wiki/Mod_event_socket
>
> I looked at a lot of examples, for me I thought the right sequence to
> would be to add (register) user agents first (determined by myself) and
> later add caller in which all user agents would ring and the call would be
> handled after accepting the call from free user agent.
>
> My problem is now the right notation of the add fifo_member. I think I
> should use this for registering agents;
>
> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>
>
> http://wiki.freeswitch.org/wiki/Mod_fifo#Simple%20On-hook%20Agent%20Login/Logout%20Example
>
> http://wiki.freeswitch.org/wiki/Variable_fifo_member_wait
>
> sendmsg <uuid>
> call-command: execute
> execute-app-name: <one of the applications>
> execute-app-arg: <application data>
>
> sendMsg
>
> call-command: execute
>
> execute-app-name: set
>
> execute-app-arg:
>
> "fifo_member(add TestQue sofia/internal/1001%123.123.123.123)"
>
> I don't know if the () can be used in sendMsg, my queue name needs quotes "\'" and the callerid needs "@" or "%" with domain at end.
>
>
> When I execute the command I don't get errors, response is ok and in log
> of freeswitch I also see no errors (only set command).
>
> However when I execute fifo list or (verbose_list) from FS CLI, I don't
> see any user agents registered. If I put the caller in my queue, I see only
> the caller registered (callers)
> http://wiki.freeswitch.org/wiki/Mod_fifo#Terminology
>
> http://wiki.freeswitch.org/wiki/Mod_fifo#list.7Clist_verbose
>
> Below works, only my created queue is missing lag, call timeout and
> simcount!
>
>  Put a caller into a FIFO queue
>
> sendMsg
>
> call-command: execute
>
> execute-app-name: fifo
>
> execute-app-arg: TestQue in somesoundfile.wav someothersoundfilemusic.wav
>
> Kind regards,
>
> Peter
>
> Ps, I hope somebody can advise !!!!
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 24 12:10:50 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 24 Sep 2012 10:10:50 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
Message-ID: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>

hi all,

at the moment we are playing arround with openVZ (container) and
freeswitch. as far as we have read, openVZ and fs are working quite
well together.

but we are getting a lot of fs core dumps, when testing. we are using
actual versions of fs, which are running very smooth on normal/real
servers without openVZ. the only information we get out of the core
dump is: "Program terminated with signal 6, Aborted."

are there any settings or things, which have to be looked at, when
running fs in an openVZ container?


thank you for your help!
dennis


From gmaruzz at gmail.com  Mon Sep 24 12:16:41 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 24 Sep 2012 10:16:41 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
Message-ID: <CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>

for me it works very well.

I used to run it on Openvz installed as rpm on a centos 6, I've
switched to a Proxmox install (that's based on Debian 6 Squeeze, and
gives you management of KVM machines too, in addition of OpenVZ
containers) and I'm very happy.

-giovanni

On Mon, Sep 24, 2012 at 10:10 AM, Dennis <odermann at googlemail.com> wrote:
> hi all,
>
> at the moment we are playing arround with openVZ (container) and
> freeswitch. as far as we have read, openVZ and fs are working quite
> well together.
>
> but we are getting a lot of fs core dumps, when testing. we are using
> actual versions of fs, which are running very smooth on normal/real
> servers without openVZ. the only information we get out of the core
> dump is: "Program terminated with signal 6, Aborted."
>
> are there any settings or things, which have to be looked at, when
> running fs in an openVZ container?
>
>
> thank you 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 odermann at googlemail.com  Mon Sep 24 12:37:03 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 24 Sep 2012 10:37:03 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
	<CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
Message-ID: <CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>

we are using proxmox too. are there some settings in proxmox or fs,
which might help? we are running ubuntu 64bit...


From gmaruzz at gmail.com  Mon Sep 24 12:46:22 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 24 Sep 2012 10:46:22 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
	<CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
	<CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>
Message-ID: <CALXCt0pF6ZVz19aWB9L_2UD6f3PepBrNgHgXtF-9cKw1zOkR7Q@mail.gmail.com>

ubuntu64 template I assume.

No special settings here, just be sure to give it (the container)
enough resources (eg: memory)

Also, we use the debian 6 template (debian 6 is used by FS dev as
platform, other platform being centos 5) but I would exclude is a
template problem.

Check the UBC counters if you got some overrun of limits.

-giovanni

On Mon, Sep 24, 2012 at 10:37 AM, Dennis <odermann at googlemail.com> wrote:
> we are using proxmox too. are there some settings in proxmox or fs,
> which might help? we are running ubuntu 64bit...
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 odermann at googlemail.com  Mon Sep 24 13:03:22 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 24 Sep 2012 11:03:22 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALXCt0pF6ZVz19aWB9L_2UD6f3PepBrNgHgXtF-9cKw1zOkR7Q@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
	<CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
	<CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>
	<CALXCt0pF6ZVz19aWB9L_2UD6f3PepBrNgHgXtF-9cKw1zOkR7Q@mail.gmail.com>
Message-ID: <CALaOL+htcrXvt87iqEAZ_EkJ4CVa9j9ANZVVu5UhLFaoE_g8Fw@mail.gmail.com>

thank you for your help!

perhaps we should try debian instead of ubuntu (although all our fs
servers are running ubuntu without any problems)?

we tested by giving the container 2 cpu cores and 2 gig of ram. do you
think the 2 gig ram are not enough, so that fs core dumps because of
this?

i don't know what ubc counters are and how they are used, but i will
try to find out.


From gmaruzz at gmail.com  Mon Sep 24 13:12:51 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 24 Sep 2012 11:12:51 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALaOL+htcrXvt87iqEAZ_EkJ4CVa9j9ANZVVu5UhLFaoE_g8Fw@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
	<CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
	<CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>
	<CALXCt0pF6ZVz19aWB9L_2UD6f3PepBrNgHgXtF-9cKw1zOkR7Q@mail.gmail.com>
	<CALaOL+htcrXvt87iqEAZ_EkJ4CVa9j9ANZVVu5UhLFaoE_g8Fw@mail.gmail.com>
Message-ID: <CALXCt0pxey5vh3xpRE=WwiP5D-w7rjJjFCyZAqZNQLVvbSd-Kw@mail.gmail.com>

On Mon, Sep 24, 2012 at 11:03 AM, Dennis <odermann at googlemail.com> wrote:
> perhaps we should try debian instead of ubuntu (although all our fs
> servers are running ubuntu without any problems)?

I would say it's irrelevant, ubuntu or debian. Crashes not from here
>
> we tested by giving the container 2 cpu cores and 2 gig of ram. do you
> think the 2 gig ram are not enough, so that fs core dumps because of
> this?

it depends on traffic, registration, etc etc etc

>
> i don't know what ubc counters are and how they are used, but i will
> try to find out.

in proxmox web interface, you have a tab UBC, bot for the host and for
the containers. You can have a look in those tabs, while containers
are running, to see if you have allocated enough resources.

Also, are you on the "stable 1.2" git branch? (
http://wiki.freeswitch.org/wiki/Installation_Guide#Selecting_version )

-giovanni

>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 wstephen80 at gmail.com  Mon Sep 24 13:17:02 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 24 Sep 2012 11:17:02 +0200
Subject: [Freeswitch-users] Remove UPDATE from Allow list in the INVITE
In-Reply-To: <CAKzWOxXN-hsFuBmURo67NeOr5csoraU-ZL1kio63x8XS9gGTiQ@mail.gmail.com>
References: <CAO4LeU1UMDY3hNcZPy7VoWSyeRhiTzURTEGy0QU4S+RGofLukg@mail.gmail.com>
	<CAKbxfG_n4iU4qmvcNf_cWvqkNxE2t7oQVDg3yiWpLvKWfoqv7w@mail.gmail.com>
	<CAO4LeU3iK0R9ruO0tXjRgB3-3pTyzR=9s0f5KuGLPNNoDGvAuA@mail.gmail.com>
	<CAKzWOxXN-hsFuBmURo67NeOr5csoraU-ZL1kio63x8XS9gGTiQ@mail.gmail.com>
Message-ID: <CAO4LeU38tn8U2Wo9pQ5yMR5phJojOZiZRAgQu6=v7XFk2USMbQ@mail.gmail.com>

I have tried this feature and it works fine. Adding the following line in
my sip profile xml config:

    <param name="send-display-update" value="false" />

the Allow list is sent without UPDATE message.

I have a question on this: why it's not possible to change this parameter
at call level? I have tried with:

     <action application="set_profile_var" data="send-display-update=false"
/>

in my dialplan but without success.


On Wed, Sep 12, 2012 at 1:57 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Sep 11, 2012 at 2:20 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Thank you Anthony, I'm updating...
>>
>> Stephen
>>
>
> Please let us know what happens! This item needs wikification once we know
> it behaves as intended.
> -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 gabe at gundy.org  Mon Sep 24 13:19:34 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 24 Sep 2012 03:19:34 -0600
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <CAMwZ-tLJjf02vACgORBebergpQssftUOU=WT=69dc_DeiunOJQ@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
	<C43E3862-427B-454D-AA1C-F59AB3F0F391@gmail.com>
	<CAMwZ-tLJjf02vACgORBebergpQssftUOU=WT=69dc_DeiunOJQ@mail.gmail.com>
Message-ID: <CAMwZ-tLXr-a0sGCC7DiM5S42ideZPyZP-vtBTueswBWL53D_sA@mail.gmail.com>

On Sun, Sep 23, 2012 at 5:07 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Sun, Sep 23, 2012 at 9:17 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
>> The part although not the same website is on;
>>
>> https://profiles.google.com/100965631603901259030/buzz
>>
>> "Another tip is to use cacheable='true' attribute of user xml returned by
>> web service. This will make FS to maintain a local cache of the account
>> details. The details will stay in cache until cleared. This means that FS
>> won't be hitting your web app for every REGISTER request that hoards of SIP
>> UAs are sending, it will ask only once upon first request. The subsequent
>> requests will be looked up from cache."
>
> Ok, I see what you're talking about. I think what you're looking at
> isn't mod_xml_curl wide. It seems to be specific to user registration.
> I'm not sure about this, but I don't see the word 'cacheable' in the
> mod_xml_curl source, but you do see it on line 1946 of switch_xml.c.
> And there too, it appears to have something to do with users.

Great... now you've got me thinking about this. I'd like to satisfy my
curiosity, but naturally, there's nothing in the wiki and no comments
in the source code :)

I wonder, what does it mean to be 'cached'? And when does that cache
get cleared? Now I must know ;)


Gabe


From avi at avimarcus.net  Mon Sep 24 13:26:16 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 24 Sep 2012 11:26:16 +0200
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <CAKV-jFxau-Ey60-j-Y58ugyGP1PktQVWXTVD1HS+q+unoiO1sw@mail.gmail.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348443615.85843.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<CAKV-jFxau-Ey60-j-Y58ugyGP1PktQVWXTVD1HS+q+unoiO1sw@mail.gmail.com>
Message-ID: <CALNZuw=32OJiC+yWSmn1WoZzfv9dhOQOe6Qx1zzOWh8kw3oGZg@mail.gmail.com>

Brian, sorry, but mod_lcr for routing internal DIDs sounds like a terrible
idea.

a) it's not DRY -- you'll have to duplicate your internal routing tables.
b) if it fails to connect to the "local" endpoint.. then if you're using
mod_lcr for the routing too, it will then try the external carriers...
which come back in and it surely won't work better than connecting directly!

-Avi


On Mon, Sep 24, 2012 at 3:08 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> Im still liking the mod_lcr idea since you should be doing it anyway for
> something like this.
> On Sep 23, 2012 7:41 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
>
>> going back to your original message,
>>
>> 1. having a separate condition for every user means that the switch will
>> have to search through hundreds of regular expressions on every call. Not
>> nice and a waste of cpu time. So, you need to look up some database instead.
>>
>> 2. after an unsuccessful bridge, you resort in voicemail. Are you sure
>> all your users want voicemail? Some will prefer the call to be unanswered,
>> and some will want to forward the call if unsuccessful. Also some will want
>> to always forward the calls. So, again you need to look up in user
>> preferences database.
>>
>> also probably it makes sense to offload the user authentication and
>> location database to an OpenSIPS or some other proxy, and let FreeSWITCH
>> deal with call routing, media and voicemail.
>>
>>
>>
>>
>> >________________________________
>> > From: D'Arcy Cain <darcy at Vex.Net>
>> >To: FreeSWITCH-users at lists.freeswitch.org
>> >Sent: Sunday, September 23, 2012 5:01 PM
>> >Subject: [Freeswitch-users] Keeping local calls local
>> >
>> >The first thing I found with the basic setup of FreeSWITCH was that
>> >calls between my own clients still went to my provider.  This seemed
>> >like a waste to me.  To handle calls coming in I generate an XML file
>> >called dialplan/public/00_Local.xml.  Here is an example entry from it.
>> >
>> ><extension name="Local_4164251212">
>> >  <condition field="destination_number" expression="^(4164251212)$">
>> >    <action application="export" data="dialed_extension=$1"/>
>> >    <action application="bind_meta_app" data="1 b s
>> execute_extension::dx XML features"/>
>> >    <action application="bind_meta_app" data="2 b s
>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>> >    <action application="bind_meta_app" data="3 b s
>> execute_extension::cf XML features"/>
>> >    <action application="bind_meta_app" data="4 b s
>> execute_extension::att_xfer XML features"/>
>> >    <action application="set" data="ringback=${us-ring}"/>
>> >    <action application="set" data="transfer_ringback=$${hold_music}"/>
>> >    <action application="set" data="call_timeout=30"/>
>> >    <action application="set" data="hangup_after_bridge=true"/>
>> >    <action application="set" data="continue_on_fail=true"/>
>> >    <action application="hash"
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>> >    <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>> >    <action application="set"
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>> >    <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>> >    <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>> >    <action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>> >    <action application="bridge" data="user/${dialed_extension}@
>> ${domain_name}"/>
>> >    <action application="answer"/>
>> >    <action application="sleep" data="1000"/>
>> >    <action application="bridge" data="loopback/app=voicemail:default
>> ${domain_name} ${dialed_extension}"/>
>> >  </condition>
>> ></extension>
>> >
>> >I am still trying to figure out exactly what everything in there does
>> >but it seems to do the job.  Calls originating outside get delivered to
>> >the correct client's phone.  Calls between clients go out to the
>> >provider and then come back in as if it was external.
>> >
>> >I worked around this by symlinking this file to the dialplan/default
>> >directory.  This works but I wonder if this is the best way to
>> >accomplish this.
>> >
>> >Side question, can I create a simple alias for all of my DIDs and make
>> >one condition that tests that alias somehow?
>> >
>> >Cheers.
>> >
>> >--
>> >D'Arcy J.M. Cain
>> >System Administrator, Vex.Net
>> >http://www.Vex.Net/ IM:darcy at Vex.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
>> >
>> >FreeSWITCH-users mailing list
>> >FreeSWITCH-users at lists.freeswitch.org
>> >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >http://www.freeswitch.org
>> >
>> >
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 24 13:28:32 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 24 Sep 2012 11:28:32 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALXCt0pxey5vh3xpRE=WwiP5D-w7rjJjFCyZAqZNQLVvbSd-Kw@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
	<CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
	<CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>
	<CALXCt0pF6ZVz19aWB9L_2UD6f3PepBrNgHgXtF-9cKw1zOkR7Q@mail.gmail.com>
	<CALaOL+htcrXvt87iqEAZ_EkJ4CVa9j9ANZVVu5UhLFaoE_g8Fw@mail.gmail.com>
	<CALXCt0pxey5vh3xpRE=WwiP5D-w7rjJjFCyZAqZNQLVvbSd-Kw@mail.gmail.com>
Message-ID: <CALaOL+gdr0vrf2AU42rD7sjab3Vpak6PjXbzG6xmhd0DVjR6uw@mail.gmail.com>

we are testing with 1.3.0 + git. this version is running withour any
problems on another fs server.

i fount the UBC tab in proxmox. failcnt is 0 and it looks, as if the
limits are set high enough.

very strange, we tested proxmox on two different servers with the same
result. so the hardware shouldn't be the problem.


From gmaruzz at gmail.com  Mon Sep 24 13:34:50 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 24 Sep 2012 11:34:50 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALaOL+gdr0vrf2AU42rD7sjab3Vpak6PjXbzG6xmhd0DVjR6uw@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
	<CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
	<CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>
	<CALXCt0pF6ZVz19aWB9L_2UD6f3PepBrNgHgXtF-9cKw1zOkR7Q@mail.gmail.com>
	<CALaOL+htcrXvt87iqEAZ_EkJ4CVa9j9ANZVVu5UhLFaoE_g8Fw@mail.gmail.com>
	<CALXCt0pxey5vh3xpRE=WwiP5D-w7rjJjFCyZAqZNQLVvbSd-Kw@mail.gmail.com>
	<CALaOL+gdr0vrf2AU42rD7sjab3Vpak6PjXbzG6xmhd0DVjR6uw@mail.gmail.com>
Message-ID: <CALXCt0pRRKxSY2Ym9L4s-PtQuNC_SGCchfiv_sxyRusDR_6MTw@mail.gmail.com>

1.3.0 is the development version, it can have problems, at times. btw,
maybe you are at a different revision between containers and other
servers.

Anyway, the policy for which code base to use is changed! Before it
was the git trunk (you are using it), now (since a couple months) is
the stable git branch. Please refer to my previous mail.

Hope this helps

-giovanni

On Mon, Sep 24, 2012 at 11:28 AM, Dennis <odermann at googlemail.com> wrote:
> we are testing with 1.3.0 + git. this version is running withour any
> problems on another fs server.
>
> i fount the UBC tab in proxmox. failcnt is 0 and it looks, as if the
> limits are set high enough.
>
> very strange, we tested proxmox on two different servers with the same
> result. so the hardware shouldn't be the 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jaybinks at gmail.com  Mon Sep 24 14:41:01 2012
From: jaybinks at gmail.com (jay binks)
Date: Mon, 24 Sep 2012 20:41:01 +1000
Subject: [Freeswitch-users] FreeSWITCH HA Database changes in 1.2
In-Reply-To: <CAKbxfG--pJq4VTrO4WPCRM_+iWCEviQo7Vg7Nvt730nF7B+V_w@mail.gmail.com>
References: <0D1C698866F66045A6201FD0F59CAC900146996AD5@EX.frontier.local>
	<CAKbxfG--pJq4VTrO4WPCRM_+iWCEviQo7Vg7Nvt730nF7B+V_w@mail.gmail.com>
Message-ID: <CAO47Hpe-0NuT418PwWhx7s=LWR95dcOFDPzms22FH_Q0i4+aMg@mail.gmail.com>

Does this now mean that we dint have to use a shared db for recovery ??

I thought that was a requirement , however maybe I missed something .
On Sep 24, 2012 10:49 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

>
>
> On Fri, Sep 21, 2012 at 6:33 PM, Colin Mason <cmason at frontiernetworks.ca>wrote:
>
>> Hello,****
>>
>> ** **
>>
>> I noticed recently on FS 1.2 that the table ?sip_recovery? wasn?t being
>> created automatically by FreeSWITCH. If I specify:****
>>
>> ** **
>>
>> <param name="core-recovery-db-dsn"
>> value="maxpowersoft_odbc:freeswitch:blahblah"/>****
>>
>> ** **
>>
>> In switch.conf.xml, FreeSWITCH will create a table called ?recovery?.****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> So my questions are:****
>>
>> ** **
>>
>> **1.       **Have you renamed the sip_recovery table to recovery?****
>>
>> **
>>
> Yes its been centralized to the core and abstracted so other endpoints can
> use it.
>
>
>
>
>
>> 2.       **Is ?core-recovery-db-dsn? a required parameter for HA
>> FreeSWITCH to function?****
>>
>> **
>>
>
> Its required if you choose to use odbc for your recovery table instead of
> sqlite.  Basically set it to the same value as odbc-dsn if that is what you
> were using with sofia, or give it a dedicated database.
>
>
>
>>  **
>>
>> Thanks,****
>>
>> Colin****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 avi at avimarcus.net  Mon Sep 24 15:02:07 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 24 Sep 2012 13:02:07 +0200
Subject: [Freeswitch-users] FreeSWITCH HA Database changes in 1.2
In-Reply-To: <CAO47Hpe-0NuT418PwWhx7s=LWR95dcOFDPzms22FH_Q0i4+aMg@mail.gmail.com>
References: <0D1C698866F66045A6201FD0F59CAC900146996AD5@EX.frontier.local>
	<CAKbxfG--pJq4VTrO4WPCRM_+iWCEviQo7Vg7Nvt730nF7B+V_w@mail.gmail.com>
	<CAO47Hpe-0NuT418PwWhx7s=LWR95dcOFDPzms22FH_Q0i4+aMg@mail.gmail.com>
Message-ID: <CALNZuwkeaHchqQat2znoF6FXC+iMAX32+Kdkb1vtTA0ud2q0BA@mail.gmail.com>

If you want to recover calls on FS crash, on the same machine, you recover
from the DB either in sqlite, whatever.
If you want to recover from a different FS instance, then of course you
need to have access to it's recovery table.

Just now you can use master-master odbc for the recovery table.. and leave
the core in sqlite or put it in a ram disk... they are separate now.

-Avi



On Mon, Sep 24, 2012 at 12:41 PM, jay binks <jaybinks at gmail.com> wrote:

> Does this now mean that we dint have to use a shared db for recovery ??
>
> I thought that was a requirement , however maybe I missed something .
> On Sep 24, 2012 10:49 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
> wrote:
>
>>
>>
>> On Fri, Sep 21, 2012 at 6:33 PM, Colin Mason <cmason at frontiernetworks.ca>wrote:
>>
>>> Hello,****
>>>
>>> ** **
>>>
>>> I noticed recently on FS 1.2 that the table ?sip_recovery? wasn?t being
>>> created automatically by FreeSWITCH. If I specify:****
>>>
>>> ** **
>>>
>>> <param name="core-recovery-db-dsn"
>>> value="maxpowersoft_odbc:freeswitch:blahblah"/>****
>>>
>>> ** **
>>>
>>> In switch.conf.xml, FreeSWITCH will create a table called ?recovery?.***
>>> *
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> So my questions are:****
>>>
>>> ** **
>>>
>>> **1.       **Have you renamed the sip_recovery table to recovery?****
>>>
>>> **
>>>
>> Yes its been centralized to the core and abstracted so other endpoints
>> can use it.
>>
>>
>>
>>
>>
>>> 2.       **Is ?core-recovery-db-dsn? a required parameter for HA
>>> FreeSWITCH to function?****
>>>
>>> **
>>>
>>
>> Its required if you choose to use odbc for your recovery table instead of
>> sqlite.  Basically set it to the same value as odbc-dsn if that is what you
>> were using with sofia, or give it a dedicated database.
>>
>>
>>
>>>  **
>>>
>>> Thanks,****
>>>
>>> Colin****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 odermann at googlemail.com  Mon Sep 24 15:11:58 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 24 Sep 2012 13:11:58 +0200
Subject: [Freeswitch-users] openVZ and FS: core dump!?
In-Reply-To: <CALXCt0pRRKxSY2Ym9L4s-PtQuNC_SGCchfiv_sxyRusDR_6MTw@mail.gmail.com>
References: <CALaOL+jSe0skA8qTAgLyRVdx28fZ3ZMNFny=G8ZhW0fiAEnwnQ@mail.gmail.com>
	<CALXCt0piXJO301WYO9X1vxrUyjF9aYP8ytaaDZJRnpsN7Tgp5Q@mail.gmail.com>
	<CALaOL+i6dWix-w5pOOm0QbeR=6FKF-X2dbqh76eGjrHd8k5AHg@mail.gmail.com>
	<CALXCt0pF6ZVz19aWB9L_2UD6f3PepBrNgHgXtF-9cKw1zOkR7Q@mail.gmail.com>
	<CALaOL+htcrXvt87iqEAZ_EkJ4CVa9j9ANZVVu5UhLFaoE_g8Fw@mail.gmail.com>
	<CALXCt0pxey5vh3xpRE=WwiP5D-w7rjJjFCyZAqZNQLVvbSd-Kw@mail.gmail.com>
	<CALaOL+gdr0vrf2AU42rD7sjab3Vpak6PjXbzG6xmhd0DVjR6uw@mail.gmail.com>
	<CALXCt0pRRKxSY2Ym9L4s-PtQuNC_SGCchfiv_sxyRusDR_6MTw@mail.gmail.com>
Message-ID: <CALaOL+jzbfBwDJDmAPohk6SUFjCs2v+o1PLU+q0-ia-XahJ+Qg@mail.gmail.com>

we nearly never use branches, because often things are not working the
way we need it or other important features are added meanwhile.
therefore we often use newer versions, we test them and if everything
is working, we use them.

perhaps we should try fs 1.2 with proxmox to make sure, there are no
problems in the newer fs versions with proxmox. but as ia wrote: we
currently running fs 1.3 without any problems on other servers.


From vitaliy.davudov at vts24.ru  Mon Sep 24 16:23:15 2012
From: vitaliy.davudov at vts24.ru (=?windows-1251?Q?=C2=E8=F2=E0=EB=E8=E9_=C4=E0=E2=F3=E4=EE=E2?=)
Date: Mon, 24 Sep 2012 16:23:15 +0400
Subject: [Freeswitch-users] Git checkout failed
In-Reply-To: <CC81DCCF.5317A%krice@freeswitch.org>
References: <CC81DCCF.5317A%krice@freeswitch.org>
Message-ID: <506050B3.1020801@vts24.ru>

Hi!

Thanksfor all your answers!

I successfully upgraded my FS. And now version is:

freeswitch at internal> version
FreeSWITCH Version 1.2.3*+**git~20120920T220849Z~f718a5e8e6* (1.2.3; git 
at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z)

Is it stable? And if it's stable, why it with *+**git...*

21.09.2012 17:58, Ken Rice ?????:
> Sounds like you might have an older version of git... Try a git fetch --all
> then a git checkout v1.2.stable
>
>
> On 9/21/12 8:42 AM, "??????? ???????" <vitaliy.davudov at vts24.ru> wrote:
>
>> Hi, list!
>>
>> I try to update my freeswitch to stable version 1.2. According to FS
>> Wiki
>> (http://wiki.freeswitch.org/wiki/Installation_Guide#Selecting_version) I
>> applied this command in previos crated freeswitch.git directory:
>>
>> git checkout v1.2.stable
>>
>> But system returned me this error:
>>
>> error: pathspec 'v1.2.stable' did not match any file(s) known to git.
>>
>> What did I do wrong?

-- 
? ?????????? ???????????,
??????? ??????? ?????????
??? "???-???????-??????"
(?????? ???????? "ETERIA")
http://www.vts24.ru
???: (495) 989-47-00

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From miconda at gmail.com  Mon Sep 24 15:38:00 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 24 Sep 2012 13:38:00 +0200
Subject: [Freeswitch-users] voicebox email notifications on Debian with exim4
Message-ID: <50604618.4010602@gmail.com>

Hello,

I got into the crashing issue for freeswitch voicemail email 
notifications on Debian with exim4, reported at:
   * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings

My solution was to use msmtp alongside exim4 -- msmtp is used to relay 
the emails to the local exim4 instance. All seems to work ok, I wonder 
if anyone else is using same approach and got any issues or there are 
other solutions found meanwhile for debian+exim4 combination.

I plan to put more details there, once the registration process for my 
account on the wiki is completed.

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
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu



From peetzer at gmail.com  Mon Sep 24 12:43:06 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Mon, 24 Sep 2012 10:43:06 +0200
Subject: [Freeswitch-users] Fifo agent status and callerid
Message-ID: <CACSMF4wxKksV5a02OLd8wN-hMmMGxB4ad4mhfSphBSiJvyABNQ@mail.gmail.com>

Hi all,

I'm implementing fifo.

I have registered my agents succesfully and drop a callerid in my queue.

The phones (agents) start to ring and show the correct callerid.

However the status I receive is from the caller and I expected this would
be from the agents (the agent is being called and not the consumer).

The caller-callerid-number should be of the agent I believe?

Should I add something like fifo_orbit_exten or fifo_consumer_id ?

Regards Peter
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From peetzer at gmail.com  Mon Sep 24 13:30:36 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Mon, 24 Sep 2012 11:30:36 +0200
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <CAMwZ-tLXr-a0sGCC7DiM5S42ideZPyZP-vtBTueswBWL53D_sA@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
	<C43E3862-427B-454D-AA1C-F59AB3F0F391@gmail.com>
	<CAMwZ-tLJjf02vACgORBebergpQssftUOU=WT=69dc_DeiunOJQ@mail.gmail.com>
	<CAMwZ-tLXr-a0sGCC7DiM5S42ideZPyZP-vtBTueswBWL53D_sA@mail.gmail.com>
Message-ID: <CACSMF4z2_jH_qhAwGC3Has6eg=-wwR-SHwRUciu8XVSu8mtOJA@mail.gmail.com>

Maybe consult with Godson G ?
Looks almost like your name ;)

Regards Peter


On Mon, Sep 24, 2012 at 11:19 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Sep 23, 2012 at 5:07 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> > On Sun, Sep 23, 2012 at 9:17 AM, Peter van Raamsdonk <peetzer at gmail.com>
> wrote:
> >> The part although not the same website is on;
> >>
> >> https://profiles.google.com/100965631603901259030/buzz
> >>
> >> "Another tip is to use cacheable='true' attribute of user xml returned
> by
> >> web service. This will make FS to maintain a local cache of the account
> >> details. The details will stay in cache until cleared. This means that
> FS
> >> won't be hitting your web app for every REGISTER request that hoards of
> SIP
> >> UAs are sending, it will ask only once upon first request. The
> subsequent
> >> requests will be looked up from cache."
> >
> > Ok, I see what you're talking about. I think what you're looking at
> > isn't mod_xml_curl wide. It seems to be specific to user registration.
> > I'm not sure about this, but I don't see the word 'cacheable' in the
> > mod_xml_curl source, but you do see it on line 1946 of switch_xml.c.
> > And there too, it appears to have something to do with users.
>
> Great... now you've got me thinking about this. I'd like to satisfy my
> curiosity, but naturally, there's nothing in the wiki and no comments
> in the source code :)
>
> I wonder, what does it mean to be 'cached'? And when does that cache
> get cleared? Now I must know ;)
>
>
> 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 miconda at gmail.com  Mon Sep 24 18:41:27 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 24 Sep 2012 16:41:27 +0200
Subject: [Freeswitch-users] Tutorial about integration of Kamailio 3.3.x and
	FreeSwitch 1.2.x
Message-ID: <50607117.3040609@gmail.com>

Hello,

just to let everyone know that I updated my tutorial about using 
Kamailio SIP Server and FreeSwitch for building telephony systems. This 
time I used latest stable branches from the two projects. The tutorial 
is available online at:
   * http://asipto.com/u/k33fs

It follows the same concept of using Kamailio for handling 
authentication, registration and routing logic, with FreeSwitch 
providing the media services (voicemail, conferencing, ivr, ...) and SBC 
functionality. One thing promised with the previous version was ability 
to generate user directory automatically from Kamailio's subscriber 
table. It is done now for voicemail using a Lua script that connects to 
Kamailio's database and builds the appropriate XML directory file.

Hope is going to be useful for many out there!

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
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu



From gmaruzz at gmail.com  Mon Sep 24 18:59:22 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 24 Sep 2012 16:59:22 +0200
Subject: [Freeswitch-users] Tutorial about integration of Kamailio 3.3.x
 and FreeSwitch 1.2.x
In-Reply-To: <50607117.3040609@gmail.com>
References: <50607117.3040609@gmail.com>
Message-ID: <CALXCt0qudxKW7WYfErLPARN8N=Q_9cXGJ=XiLQ8eGT=fjmncmQ@mail.gmail.com>

I was experimenting right with database views!

Thanks Daniel Constantin!

-giovanni


On 9/24/12, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
> Hello,
>
> just to let everyone know that I updated my tutorial about using
> Kamailio SIP Server and FreeSwitch for building telephony systems. This
> time I used latest stable branches from the two projects. The tutorial
> is available online at:
>    * http://asipto.com/u/k33fs
>
> It follows the same concept of using Kamailio for handling
> authentication, registration and routing logic, with FreeSwitch
> providing the media services (voicemail, conferencing, ivr, ...) and SBC
> functionality. One thing promised with the previous version was ability
> to generate user directory automatically from Kamailio's subscriber
> table. It is done now for voicemail using a Lua script that connects to
> Kamailio's database and builds the appropriate XML directory file.
>
> Hope is going to be useful for many out there!
>
> 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
> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -
> http://asipto.com/u/katu
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 the_spide21 at yahoo.com  Mon Sep 24 19:12:25 2012
From: the_spide21 at yahoo.com (Carlos Lopez)
Date: Mon, 24 Sep 2012 08:12:25 -0700 (PDT)
Subject: [Freeswitch-users] Automatic Callout and TTS
In-Reply-To: <EDF753B814C44E44BD108AE6DCEF73E8@e1705>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<EDF753B814C44E44BD108AE6DCEF73E8@e1705>
Message-ID: <1348499545.93808.YahooMailNeo@web45107.mail.sp1.yahoo.com>

Hi all,

I am just learning how to install and use Freeswitch and at the place where I'm working i'd like to implement a solution like callout and TTS.

The main goal is to use Freeswith as a PBX and use text based strings so the FSW can callout some customers and advise them about theit current balance so the string will be componded by a set of fields and the resulting text most be converted to phone sound to the customer, ex.

1- Event occurs on our main system server.
2- Server send a sms/E-Mail/HTTP-String to FSW, with customer phone # and string ( String will be like: "You have % credit balance on your account. Please type 1 to add credit", where % is a numeric value that will be filled up by our main system servers.
3- FSW will convert resulting text to sound call and calls the customer's phone number.
4- Customer picks up the phone and listen to the TTS sound.

Is it possible with FSW?

Thank you all for your response.

Carlos.
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From yehavi.bourvine at gmail.com  Mon Sep 24 19:56:13 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 24 Sep 2012 17:56:13 +0200
Subject: [Freeswitch-users] Softphone with callee name display?
In-Reply-To: <CALn233q=w2NvBxm+Hzo5VcOr3h1fwAWPGbkC_gUVq=KMupa+vg@mail.gmail.com>
References: <CAAtWKeHznz7bLSMnLJ_tn=BZH=mFPO5NDb9S0q0kWWZ8GrU1nA@mail.gmail.com>
	<CALn233q=w2NvBxm+Hzo5VcOr3h1fwAWPGbkC_gUVq=KMupa+vg@mail.gmail.com>
Message-ID: <CAAtWKeFEP456Gqsyh=V=SV9L7E1-M18-2mXSyjY-eOYjearc7Q@mail.gmail.com>

I am looking for the called name, not caller name...
              Thanks, __Yehavi:
2012/9/24 Mitch Capper <mitch.capper at gmail.com>

> FSClient shows caller id:)
>
> ~Mitch
>
> On Mon, Sep 24, 2012 at 5:53 AM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
> > Hello,
> >
> >   Polycom and Snom (at least) can display the name of the called person
> > (sent by P-Remote-ID or P-Asserted-ID fields). Is there a softphone that
> > does the same? I couldn't find suh one so far...
> >
> >                   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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Sep 24 20:00:55 2012
From: james.bravo at redmatter.com (James Bravo)
Date: Mon, 24 Sep 2012 17:00:55 +0100
Subject: [Freeswitch-users] Invalid length field in RTCP Sender Report
In-Reply-To: <501E5CAE.6080305@redmatter.com>
References: <501BC9F9.3070703@redmatter.com> <501E40B3.7030106@elder.hu>
	<501E5CAE.6080305@redmatter.com>
Message-ID: <506083B7.1020709@redmatter.com>

Hi

We've had a number of Yealink T22 & T28 phones with later
firmware crashing within approx 30 seconds of sending
and receiving RTP.
Yealink R&D department have narrowed it down to a
non-standard RTCP 'Sender Report' packet from FreeSWITCH.
For some packet content, the length field in these packets
does not seem to agree with the overall UDP packet size.
RFC1889 defines the length as:
     "The length of this RTCP packet in 32-bit words minus one,
     including the header and any padding."
For instance, below is a RTCP packet sent from FS as viewed
in Wireshark. The length field says 76 bytes but the udp packet
length is 78 bytes so the last two bytes (0x31, 0x00) are excluded
as far as the length field is concerned.


I managed to stop the phones crashing by changing the following lines
in switch_rtp.c...
         rtcp_bytes = sizeof(switch_rtcp_hdr_t) + sizeof(struct 
switch_rtcp_senderinfo) + sr->sr_desc_ssrc.length -1 ;
         rtp_session->rtcp_send_msg.header.length = 
htons((u_short)(rtcp_bytes / 4) - 1);
...to something like...
     {
         switch_size_t rtcp_bytes_mod4;
         rtcp_bytes = sizeof(switch_rtcp_hdr_t) + sizeof(struct 
switch_rtcp_senderinfo) + sr->sr_desc_ssrc.length -1 ;
         // Add padding and adjust length if necessary
         rtcp_bytes_mod4 = rtcp_bytes % 4;
         if (rtcp_bytes_mod4 > 0)
             rtcp_bytes += 4 - rtcp_bytes_mod4;
         rtp_session->rtcp_send_msg.header.length = 
htons((u_short)(rtcp_bytes / 4 - 1));
     }
although Freeswitch developers will probably find a better solution.

Thanks in advance,
James Bravo,
Redmatter

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From krice at freeswitch.org  Mon Sep 24 20:07:32 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 24 Sep 2012 11:07:32 -0500
Subject: [Freeswitch-users] Invalid length field in RTCP Sender Report
In-Reply-To: <506083B7.1020709@redmatter.com>
Message-ID: <CC85EF74.5326F%krice@freeswitch.org>

This sort of thing should be reported via Jira ( http://jira.freeswitch.org
) will any logs, and patches attached...


On 9/24/12 11:00 AM, "James Bravo" <james.bravo at redmatter.com> wrote:

>    Hi
>  
>  We've had a number of Yealink T22 & T28 phones with later
>  firmware crashing within approx 30 seconds of sending
>  and receiving RTP.
>  Yealink R&D department have narrowed it down to a
>  non-standard RTCP 'Sender Report' packet from FreeSWITCH.
>  For some packet content, the length field in these packets
>  does not seem to agree with the overall UDP packet size.
>  RFC1889 defines the length as:
>      "The length of this RTCP packet in 32-bit words minus one,
>      including the header and any padding."
>  For instance, below is a RTCP packet sent from FS as viewed
>  in Wireshark. The length field says 76 bytes but the udp packet
>  length is 78 bytes so the last two bytes (0x31, 0x00) are excluded
>  as far as the length field is concerned.
>  
>  
>  I managed to stop the phones crashing by changing the following lines
>  in switch_rtp.c...
>          rtcp_bytes = sizeof(switch_rtcp_hdr_t) + sizeof(struct
> switch_rtcp_senderinfo) + sr->sr_desc_ssrc.length -1 ;
>          rtp_session->rtcp_send_msg.header.length = htons((u_short)(rtcp_bytes
> / 4) - 1); 
>  ...to something like...
>      {
>          switch_size_t rtcp_bytes_mod4;
>          rtcp_bytes = sizeof(switch_rtcp_hdr_t) + sizeof(struct
> switch_rtcp_senderinfo) + sr->sr_desc_ssrc.length -1 ;
>          // Add padding and adjust length if necessary
>          rtcp_bytes_mod4 = rtcp_bytes % 4;
>          if (rtcp_bytes_mod4 > 0)
>              rtcp_bytes += 4 - rtcp_bytes_mod4;
>          rtp_session->rtcp_send_msg.header.length = htons((u_short)(rtcp_bytes
> / 4 - 1)); 
>      }
>  although Freeswitch developers will probably find a better solution.
>  
>  Thanks in advance,
>  James Bravo, 
>  Redmatter
>  
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mario_fs at mgtech.com  Mon Sep 24 20:47:18 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 24 Sep 2012 09:47:18 -0700
Subject: [Freeswitch-users] FreeSwitch Video Calls Bria iPhone Client
In-Reply-To: <7DE0AD3C-2A16-49B6-8533-3F0834D54768@me.com>
References: <2B4CD419-FE1B-443C-86AB-2E5A98C63ED2@me.com>
	<603F099C-8BDB-4EB5-9375-6DEF87CF06EE@mgtech.com>
	<CAO47HpeOng3U4TE8TPLY6JcAyTvJYiufYpfUT30qz3OgNqhLNA@mail.gmail.com>
	<7DE0AD3C-2A16-49B6-8533-3F0834D54768@me.com>
Message-ID: <9D3AEDA9-4587-4A01-9A53-966BECD7EA60@mgtech.com>

Thanks for the response!  I have tested without ALG months ago to no avail. No problems for over a year, it started when I upgraded FreeSwitch from early 2011 to early 2012 version. However, at the same time added the iPads/Bria and switched the Linksys SPA962s from SLA to individual user IDs. I am suspecting the Linksys SPA962s doing something wonky (and affecting Bria?) which is why I am replacing them. Wanted to go all iPads, especially since Apple may come out with a 7 incher soon. Otherwise looking at SNOM 870s and Polycoms. This is driving me nuts since some days no problem and some days it's bad with no pattern. Lots of traces.

FreeSwitch is connected to the Zyxel using multiple WANs (static + dynamic) which balance and backup each other. Works great, can pull the plug on either at any time and FreeSwitch goes merrily along and recovers (when I get this problem sorted out I will add a wiki about the setup). Using the router FreeSwitch knows nothing about the IPs, NAT, etc. no messy setup. SIP ALG is always suspected first since it gets such a bad rap but so far no issues here. The problems below don't seem to be related to it. If I ever get this sorted out I will post in case some other poor soul runs into this. Thanks again! Wish I could have helped with your issue.
Mario G

On Sep 23, 2012, at 6:59 PM, Mike Burlingame wrote:

> I agree I am not seeing any issues on call setup / take down or audio RTP or SMS only issues with the video setting up correctly
> 
> Have you tried to remove the ALG and reproduced the same issue?
> 
> Sent from my iPhone 4S
> 
> On Sep 23, 2012, at 5:39 PM, jay binks <jaybinks at gmail.com> wrote:
> 
>> You know your zyxel  could be the issue  for lots of that .  Sip aware Nat router = sip ALG = black magic going on that neither end can predict.
>> 
>> Remove or disable the alg  and try again .
>> 
>> On Sep 24, 2012 4:10 AM, "Mario G" <mario_fs at mgtech.com> wrote:
>> FWIW, I was trying to replace SPA962s with Bria on iPads in our office. Had several issues, at least 6 open problems with Counterpath, they were great at trying to help but pointed the finger at FreeSwitch. I posted here for months to no avail so I assume no one here using Bria to this extent. FYI, all my issues are ringing/answering related (listed below). If you find anything please post here as I plan t do the same to keep someone else from pulling their hair out.
>> 
>> 1. Bria does not always ring the iPad so you can't answer. I am using TCP to avoid battery drain although UDP did not seem to help.
>> 2. Bria rings, but when you hit answer there is nothing there and the other extensions keep ringing. FS trace shows FS disconnecting.
>> 3. If Bria is not in the foreground and a ringing alert is displayed, pressing it sometimes results in the last issue of not answering (FS says hangup) but other phones ring.
>> 4. Not a Bria or FS issue but a pain: Apparently Apple limits the alert to 25 seconds, if you don't press it in that time it closed and you can't answer. This is an IOS issue that Apple needs to allow more flexibility with.
>> 
>> Once a call is connected there are no issues with the recent Bria iPad updates, there were some issues a month ago. I highly recommend opening an issue with Counterpath as they seemed to know SIP and understood some of the FS trace stuff. They are very quick to update the Bria if an issue is found. But in my case, after 6+ months of traces we can't find the culprit.
>> 
>> I have no Nat issues at all using Zyxel routers with SIP support. Good luck to you, I am writing the president of CounterPath to see if they can get a FreeSwitch to test with since it would be great for both to work together.
>> Mario G
>> 
>> On Sep 22, 2012, at 7:46 PM, Mike Burlingame wrote:
>> 
>> > I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android?
>> >
>> > 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 msc at freeswitch.org  Mon Sep 24 20:51:22 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Sep 2012 09:51:22 -0700
Subject: [Freeswitch-users] using spandsp fax tone detection from
 originate command
In-Reply-To: <CAE2A9TBRKxwXOKHH3WwadCt+DUJ0M-sqR5yVWnK2ziP-DitxQg@mail.gmail.com>
References: <CAE2A9TDoXMEhALC5dwBd+1m8X-D2pr278nz-opgLvwVu2MBWZA@mail.gmail.com>
	<CAKzWOxWGCXkKPSvznMxmkEARxYvGSLWf2GQJ8Beds+C7+W+z7A@mail.gmail.com>
	<CAE2A9TBRKxwXOKHH3WwadCt+DUJ0M-sqR5yVWnK2ziP-DitxQg@mail.gmail.com>
Message-ID: <CAKzWOxWf7DN-O-j2aamcc36KybzbibYH2NqCwuEJJPFROhD1+Q@mail.gmail.com>

Maybe you could clean things up a bit and use an extension that sets some
of these variables:
http://wiki.freeswitch.org/wiki/Mod_spandsp#Execute_based_on_fax_session_outcome

Maybe do something like:

originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234handle_fax

Then create that extension:
<extension name="Handle faxes">
  <condition field="destination_number" expression="^handle_fax$">
    <action application="answer"/>
    <action application="execute_on_fax_detect=txfax /tmp/myfax.tif"/>
    <action application="execute_on_fax_success=perl fax_sent.pl"/>
    <action application="execute_on_fax_failure=perl fax_failed.pl"/>
    <action application="playback" data="silence_stream://20000"/>
    <action application="hangup"/>
  </condition>
</extension>

Of course, do your own thing with the execute_on_fax_xxx variables.
Standard disclaimer applies - test it and tinker before telling me it
doesn't work. :)

-MC

On Fri, Sep 21, 2012 at 12:31 PM, Adam Johnston
<ajohnston at blimessaging.com>wrote:

> I had actually added some sleep commands in some of my other command
> combinations but I kept getting the same result. However, once I increased
> the timeout to spandsp_start_fax_detect and increased the sleep time I
> started getting some successful faxes out.
>
> If no fax tone is detected I would like to set fax_result_code to a custom
> error code that I could then map in my controller application. Is there
> some way to evaluate the result of a tone detect? I've been trying the
> below inline dialplan but it's giving my false negatives so I'm back on the
> wiki looking for a solution.
>
> 'spandsp_start_fax_detect:txfax /tmp/myfax.tif 10
> ced,sleep:10000,set:fax_result_code:951' inline
>
> Many thanks,
>
> Adam Johnston
>
>
> On Fri, Sep 21, 2012 at 2:07 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Hi Adam,
>>
>> The key piece of information is this:
>>
>> [NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has
>> executed the last dialplan instruction, hanging up.
>>
>> You've enabled "fax detection" on the call but you haven't told the call
>> to actually do anything. From what you're doing it seems like you're
>> wanting to detect a fax and then run the txfax application. But what
>> happens if a fax is not detected? Also, what happens to the call in the
>> interim, i.e. between the time you enable fax detect and the time it
>> actually detects a fax tone? The call needs to do "something" while waiting
>> for a fax tone. You may be better off sending this call to an extension and
>> then having several dialplan actions, like "answer",
>> "spandsp_start_fax_detect", and then a "sleep" or "playback" app, or
>> something for the channel to do while it's waiting for fax tone.
>>
>> Question: what do you want the call to do if no fax is detected?
>>
>> -MC
>>
>> On Fri, Sep 21, 2012 at 10:26 AM, Adam Johnston <
>> ajohnston at blimessaging.com> wrote:
>>
>>> Hi all,
>>>
>>> I've been trying to use the spandsp_start_fax_detect application (with
>>> txfax as the callback) from the originate command but I keep getting a
>>> hangup before it completes.
>>>
>>> So far I've tried calling the app directly:
>>> originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234&spandsp_start_fax_detect('txfax /tmp/myfax.tif 3 ced')
>>>
>>> Inline dialplan:
>>> originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234'spandsp_start_fax_detect:txfax /tmp/myfax.tif 3 ced' inline
>>>
>>> ..plus some other permutations but all end with the same result:
>>>
>>> [NOTICE] mod_spandsp.c:107 Enabling fax detection 'txfax'
>>> '/tmp/myfax.tif'
>>> [DEBUG] switch_core_media_bug.c:456 Attaching BUG to sofia/external/
>>> 18005551234
>>> [NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has
>>> executed the last dialplan instruction, hanging up.
>>>
>>> Am I misunderstanding how spandsp_start_fax_detect works?
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 miconda at gmail.com  Mon Sep 24 22:16:50 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 24 Sep 2012 20:16:50 +0200
Subject: [Freeswitch-users] Tutorial about integration of Kamailio 3.3.x
 and FreeSwitch 1.2.x
In-Reply-To: <CALXCt0qudxKW7WYfErLPARN8N=Q_9cXGJ=XiLQ8eGT=fjmncmQ@mail.gmail.com>
References: <50607117.3040609@gmail.com>
	<CALXCt0qudxKW7WYfErLPARN8N=Q_9cXGJ=XiLQ8eGT=fjmncmQ@mail.gmail.com>
Message-ID: <5060A392.5090501@gmail.com>


On 9/24/12 4:59 PM, Giovanni Maruzzelli wrote:
> I was experimenting right with database views!
Indeed, that can be an alternative.
>
> Thanks Daniel Constantin!

Welcome,
Daniel

>
> -giovanni
>
>
> On 9/24/12, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>> Hello,
>>
>> just to let everyone know that I updated my tutorial about using
>> Kamailio SIP Server and FreeSwitch for building telephony systems. This
>> time I used latest stable branches from the two projects. The tutorial
>> is available online at:
>>     * http://asipto.com/u/k33fs
>>
>> It follows the same concept of using Kamailio for handling
>> authentication, registration and routing logic, with FreeSwitch
>> providing the media services (voicemail, conferencing, ivr, ...) and SBC
>> functionality. One thing promised with the previous version was ability
>> to generate user directory automatically from Kamailio's subscriber
>> table. It is done now for voicemail using a Lua script that connects to
>> Kamailio's database and builds the appropriate XML directory file.
>>
>> Hope is going to be useful for many out there!
>>
>> 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
>> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -
>> http://asipto.com/u/katu
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu



From miconda at gmail.com  Mon Sep 24 22:24:22 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 24 Sep 2012 20:24:22 +0200
Subject: [Freeswitch-users] voicebox email notifications on Debian with
	exim4
In-Reply-To: <50604618.4010602@gmail.com>
References: <50604618.4010602@gmail.com>
Message-ID: <5060A556.5040201@gmail.com>

I added some guidelines about using msmtp as local relay to exim4 on 
debian as an workaround for the crash:
   * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
   * 
http://wiki.freeswitch.org/wiki/Mod_voicemail#Using_MSMTP_for_Local_Relay_to_Exim4_on_Debian

Hopefully others can test to add new comments and/or improve the content.

Cheers,
Daniel


On 9/24/12 1:38 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> I got into the crashing issue for freeswitch voicemail email 
> notifications on Debian with exim4, reported at:
>   * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>
> My solution was to use msmtp alongside exim4 -- msmtp is used to relay 
> the emails to the local exim4 instance. All seems to work ok, I wonder 
> if anyone else is using same approach and got any issues or there are 
> other solutions found meanwhile for debian+exim4 combination.
>
> I plan to put more details there, once the registration process for my 
> account on the wiki is completed.
>
> 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
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu



From msc at freeswitch.org  Mon Sep 24 22:34:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Sep 2012 11:34:14 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVmHXdsRaPmJvwr1wP31uig2t5Ee9vq-uThhQuGYGrtdw@mail.gmail.com>

Welcome to the last Monday of September 2012!

We've had quite the interesting week. Perhaps the most interesting item the
team dealt with was a vulnerability in the Sofia SIP stack that would cause
a segmentation fault while processing a specially crafted SIP message. Just
to show you how nimble the FreeSWITCH developers are, from the time the
vulnerability was reported it took less than a day to fix, test, and roll a
new version of FreeSWITCH. We encourage everyone on 1.2 to get updated
to version
1.2.3 <http://files.freeswitch.org/freeswitch-1.2.3.tar.bz2> as soon as
possible. (The fix is also in the 1.3 development branch as of last
Wednesday, September 19.) We tip our hats to Anthony and the rest of the dev
team<http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_So_where_do_I_find_wishlists_for_the_developers_of_this_awesome_software.3F>for
their hard work on our behalf.

Last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19>was also
very informative. We received an introduction to the repro
SIP proxy <http://resiprocate.org/About_Repro> software. We look
forward to this
coming Wednesday
<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26>where Scott
Godin and Daniel Pocock will continue the discussion and will
get deeper into how to set up the proxy and use it with FreeSWITCH. If you
haven't already tried to install repro please do so. Daniel has a nice
tutorial<http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch>over
at OpenTelecoms.org - be sure to check it out and bring your questions
on Wednesday.

Finally, we'd like to draw your attention to this blog
post<http://blog.krisk.org/2012/09/freeswitch-stable-branch-sponsored-by.html>by
long time FreeSWITCH and open source telephony supporter Kristian
Kielhofner. Kristian reports that his company, Star2Star
Communications<http://www.star2star.com/>,
is sponsoring the FreeSWITCH stable branch by giving direct financial
support to the project. This allows for a full-time team member to work on
things like the stable branch and packaging as well as community
interaction and documentation. We appreciate those who support FreeSWITCH
and open source telephony!

Have a good week and we'll see you again in October.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From darcy at Vex.Net  Mon Sep 24 21:52:16 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Mon, 24 Sep 2012 13:52:16 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
	<1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>
	<CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>
Message-ID: <20120924135216.588785857c2c8fab2d014e55@Vex.Net>

On Mon, 24 Sep 2012 00:40:37 +0200
Avi Marcus <avi at avimarcus.net> wrote:
> enum or a DB query to your routing table for inbound calls, e.g. using
> mod_odbc_query <http://wiki.freeswitch.org/wiki/Mod_odbc_query>.

I considered this but balked at the idea of using ODBC.  I didn't want
to add another layer.  I am considering creating an application in
Python to connect to my database and get all the needed details.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From darcy at Vex.Net  Mon Sep 24 21:55:11 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Mon, 24 Sep 2012 13:55:11 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <C067307A-A078-44DE-A922-37F436C3AA36@insensate.co.uk>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<C067307A-A078-44DE-A922-37F436C3AA36@insensate.co.uk>
Message-ID: <20120924135511.7164ebb726ac4266591be1ee@Vex.Net>

On Sun, 23 Sep 2012 22:54:53 +0100
Lawrence Conroy <lconroy at insensate.co.uk> wrote:

> Hi there,
>  full disclosure -- on ENUM, I'm biased :).

Not having much joy with enum.  I tried the CLI example and got nothing
back.  I also tried some other numbers including ones suggested on the
e164.org site with no better luck.  I then tried one of my own numbers
and FreeSWITCH crashed with a socket error.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From msc at freeswitch.org  Mon Sep 24 23:01:05 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Sep 2012 12:01:05 -0700
Subject: [Freeswitch-users] Git checkout failed
In-Reply-To: <506050B3.1020801@vts24.ru>
References: <CC81DCCF.5317A%krice@freeswitch.org> <506050B3.1020801@vts24.ru>
Message-ID: <CAKzWOxXD_Vp1g+JnpoEz+Zk=Mpe_DMd34Z2SFb_CobX9aJShVA@mail.gmail.com>

On Mon, Sep 24, 2012 at 5:23 AM, ??????? ???????
<vitaliy.davudov at vts24.ru>wrote:

>  Hi!
>
> Thanks for all your answers!
>
> I successfully upgraded my FS. And now version is:
>
> freeswitch at internal> version
> FreeSWITCH Version 1.2.3*+**git~20120920T220849Z~f718a5e8e6* (1.2.3; git
> at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z)
>
> Is it stable? And if it's stable, why it with *+**git...*
>

This is more for convenience than anything else. You could just as well
download the source tarballs, but being on git means that when 1.2.4 comes
out you just need to "make current" and you'll be up to speed. (Note: this
is assuming you have actually checked out the v1.2.stable branch.)

-MC
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From msc at freeswitch.org  Mon Sep 24 23:02:15 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Sep 2012 12:02:15 -0700
Subject: [Freeswitch-users] Fifo agent status and callerid
In-Reply-To: <CACSMF4wxKksV5a02OLd8wN-hMmMGxB4ad4mhfSphBSiJvyABNQ@mail.gmail.com>
References: <CACSMF4wxKksV5a02OLd8wN-hMmMGxB4ad4mhfSphBSiJvyABNQ@mail.gmail.com>
Message-ID: <CAKzWOxVbnzYhAuQ85wofo+2PWVatTa3uDf4e7f96e0ycUfB38Q@mail.gmail.com>

Where are you receiving this status? Is it via event socket? If so, can you
pastebin an example of the whole message?
-MC

On Mon, Sep 24, 2012 at 1:43 AM, Peter van Raamsdonk <peetzer at gmail.com>wrote:

> Hi all,
>
> I'm implementing fifo.
>
> I have registered my agents succesfully and drop a callerid in my queue.
>
> The phones (agents) start to ring and show the correct callerid.
>
> However the status I receive is from the caller and I expected this would
> be from the agents (the agent is being called and not the consumer).
>
> The caller-callerid-number should be of the agent I believe?
>
> Should I add something like fifo_orbit_exten or fifo_consumer_id ?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Sep 24 23:08:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Sep 2012 12:08:27 -0700
Subject: [Freeswitch-users] Automatic Callout and TTS
In-Reply-To: <1348499545.93808.YahooMailNeo@web45107.mail.sp1.yahoo.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<EDF753B814C44E44BD108AE6DCEF73E8@e1705>
	<1348499545.93808.YahooMailNeo@web45107.mail.sp1.yahoo.com>
Message-ID: <CAKzWOxU-uSLSaspqxqHw5is3WEOv59bziHmt8T1YPJdS-=qGfQ@mail.gmail.com>

Yep, this is totally possible. First question: what programming/technical
resources do you have in house? That will determine if you need "regular"
help or "professional" help. :)

-MC

On Mon, Sep 24, 2012 at 8:12 AM, Carlos Lopez <the_spide21 at yahoo.com> wrote:

> Hi all,
>
> I am just learning how to install and use Freeswitch and at the place
> where I'm working i'd like to implement a solution like callout and TTS.
>
> The main goal is to use Freeswith as a PBX and use text based strings so
> the FSW can callout some customers and advise them about theit current
> balance so the string will be componded by a set of fields and the
> resulting text most be converted to phone sound to the customer, ex.
>
> 1- Event occurs on our main system server.
> 2- Server send a sms/E-Mail/HTTP-String to FSW, with customer phone # and
> string ( String will be like: "You have % credit balance on your account.
> Please type 1 to add credit", where % is a numeric value that will be
> filled up by our main system servers.
> 3- FSW will convert resulting text to sound call and calls the customer's
> phone number.
> 4- Customer picks up the phone and listen to the TTS sound.
>
> Is it possible with FSW?
>
> Thank you all for your response.
>
> Carlos.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 onyeagbaikenna04 at gmail.com  Mon Sep 24 23:17:23 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Mon, 24 Sep 2012 19:17:23 +0000
Subject: [Freeswitch-users] Apache
Message-ID: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>

Hi everyone
How can I get freeswitch running on apache 2.. 
Thanks for your anticipated response

Sent from my BlackBerry smartphone from Virgin Media


From onyeagbaikenna04 at gmail.com  Mon Sep 24 23:18:43 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Mon, 24 Sep 2012 19:18:43 +0000
Subject: [Freeswitch-users] Freeswitch on Apache
Message-ID: <909083812-1348514189-cardhu_decombobulator_blackberry.rim.net-2036827440-@b12.c16.bise7.blackberry>

Hi everyone
How can I get freeswitch working on apache 2..
Thanks for your anticipated response
Sent from my BlackBerry smartphone from Virgin Media


From Hector.Geraldino at ipsoft.com  Mon Sep 24 23:17:36 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Mon, 24 Sep 2012 15:17:36 -0400
Subject: [Freeswitch-users] Automatic Callout and TTS
In-Reply-To: <1348499545.93808.YahooMailNeo@web45107.mail.sp1.yahoo.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<EDF753B814C44E44BD108AE6DCEF73E8@e1705>
	<1348499545.93808.YahooMailNeo@web45107.mail.sp1.yahoo.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD07405FC3F9@NY1-EXMB-01.ip-soft.net>

You will need a man-in-the-middle application (broker) that receives the message and converts it to actionable steps that will be performed in FS.

Of the many ways you have to do this, I can suggest you to explore the Event Socket Outbound ? and particulary the Java ESL Client, although you can use pretty much any other language supported by FS. Main idea is: your application will receive the message, and using ESL will generate an outbound call using FreeSWITCH. When the call is answered it will play a message using a TTS engine (flite, cepstral, nuance, etc.), capture the input and do whatever you need with it.

The control of the call can be handled via ESL or using a dialplan.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos Lopez
Sent: Monday, September 24, 2012 11:12 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Automatic Callout and TTS

Hi all,

I am just learning how to install and use Freeswitch and at the place where I'm working i'd like to implement a solution like callout and TTS.

The main goal is to use Freeswith as a PBX and use text based strings so the FSW can callout some customers and advise them about theit current balance so the string will be componded by a set of fields and the resulting text most be converted to phone sound to the customer, ex.

1- Event occurs on our main system server.
2- Server send a sms/E-Mail/HTTP-String to FSW, with customer phone # and string ( String will be like: "You have % credit balance on your account. Please type 1 to add credit", where % is a numeric value that will be filled up by our main system servers.
3- FSW will convert resulting text to sound call and calls the customer's phone number.
4- Customer picks up the phone and listen to the TTS sound.

Is it possible with FSW?

Thank you all for your response.

Carlos.
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From krice at freeswitch.org  Mon Sep 24 23:20:03 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 24 Sep 2012 14:20:03 -0500
Subject: [Freeswitch-users] Apache
In-Reply-To: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>
Message-ID: <CC861C93.53293%krice@freeswitch.org>

Freeswitch doesn't use apache outside of libapr... Apache2 would be a
webserver... 


On 9/24/12 2:17 PM, "onyeagbaikenna04 at gmail.com"
<onyeagbaikenna04 at gmail.com> wrote:

> Hi everyone
> How can I get freeswitch running on apache 2..
> Thanks for your anticipated response
> 


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From tnsampaio at bsd.com.br  Mon Sep 24 23:43:01 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Mon, 24 Sep 2012 16:43:01 -0300
Subject: [Freeswitch-users] Recording problem speed and noise
Message-ID: <5060B7C5.1060003@bsd.com.br>

Hi All!

Im facing a strange issue..
My recordings (in mp3, wav or gsm) are too fast..
now recordings are completely inaudible, only with noise and voice 
metallized...
But sometimes it not happen, just speed issue...

im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2)

today i tryed to change to 1.3.0, but faced another problem, noise on calls.
i  checked load on my machine, and load was 0.2, almost 100% cpu FREE..
i have a khomp board with R2 signaling..

Can you help me to debug what is happening?
Im quite new to FS!!!

Hugs


From ssinyagin at yahoo.com  Tue Sep 25 00:04:07 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Mon, 24 Sep 2012 13:04:07 -0700 (PDT)
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <20120924135216.588785857c2c8fab2d014e55@Vex.Net>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
	<1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>
	<CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>
	<20120924135216.588785857c2c8fab2d014e55@Vex.Net>
Message-ID: <1348517047.51778.YahooMailNeo@web39302.mail.mud.yahoo.com>

you probably don't need to program it in Python:
Daniel-Constantin wrote a nice guide where a Lua script generates per-user XML on the fly from an SQL database:
http://asipto.com/u/k33fs





----- Original Message -----
> From: D'Arcy Cain <darcy at Vex.Net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Monday, September 24, 2012 7:52 PM
> Subject: Re: [Freeswitch-users] Keeping local calls local
> 
> On Mon, 24 Sep 2012 00:40:37 +0200
> Avi Marcus <avi at avimarcus.net> wrote:
>>  enum or a DB query to your routing table for inbound calls, e.g. using
>>  mod_odbc_query <http://wiki.freeswitch.org/wiki/Mod_odbc_query>.
> 
> I considered this but balked at the idea of using ODBC.? I didn't want
> to add another layer.? I am considering creating an application in
> Python to connect to my database and get all the needed details.
> 
> -- 
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From avi at avimarcus.net  Tue Sep 25 00:22:11 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 24 Sep 2012 22:22:11 +0200
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <1348517047.51778.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
	<1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>
	<CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>
	<20120924135216.588785857c2c8fab2d014e55@Vex.Net>
	<1348517047.51778.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <CALNZuwnab8j+px_2Fu5XVeSnZU_fWuUV6w2pn9Rupvf9OOH=CQ@mail.gmail.com>

... python is also an extra layer. Odbc query should be a high performance
interpreter-less mod.
I use it to route and rate inbound public calls.

-Avi

On Sep 24, 2012 10:06 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:

you probably don't need to program it in Python:
Daniel-Constantin wrote a nice guide where a Lua script generates per-user
XML on the fly from an SQL database:
http://asipto.com/u/k33fs






----- Original Message -----
> From: D'Arcy Cain <darcy at Vex.Net>
> To: FreeSWITCH Users Help <f...

> Sent: Monday, September 24, 2012 7:52 PM
> Subject: Re: [Freeswitch-users] Keeping local calls loc...

> _________________________________________________________________________
> Professional FreeSWITC...
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From peetzer at gmail.com  Mon Sep 24 23:13:12 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Mon, 24 Sep 2012 21:13:12 +0200
Subject: [Freeswitch-users] Fifo agent status and callerid
In-Reply-To: <CAKzWOxVbnzYhAuQ85wofo+2PWVatTa3uDf4e7f96e0ycUfB38Q@mail.gmail.com>
References: <CACSMF4wxKksV5a02OLd8wN-hMmMGxB4ad4mhfSphBSiJvyABNQ@mail.gmail.com>
	<CAKzWOxVbnzYhAuQ85wofo+2PWVatTa3uDf4e7f96e0ycUfB38Q@mail.gmail.com>
Message-ID: <5ABE82AB-0DBB-4737-A55F-6E97AC5EE652@gmail.com>

Hi Michael,

It is with event socket. Somehow I get the callerid & caller name but actually need the calleeid.
I added also the ring_ready attribute, to receive the answering of the call.

I will experiment tomorrow a bit, I thought all agents that are run with the caller-destination-number.

It is sometimes tricky what parameters to choose from either direct calling (extension to extension), outbound calling or bridging from Fifo.

Regards Peter

On 24 sep 2012, at 21:02, Michael Collins wrote:

> Where are you receiving this status? Is it via event socket? If so, can you pastebin an example of the whole message?
> -MC
> 
> On Mon, Sep 24, 2012 at 1:43 AM, Peter van Raamsdonk <peetzer at gmail.com> wrote:
> Hi all,
>  
> I'm implementing fifo.
>  
> I have registered my agents succesfully and drop a callerid in my queue.
>  
> The phones (agents) start to ring and show the correct callerid.
>  
> However the status I receive is from the caller and I expected this would be from the agents (the agent is being called and not the consumer).
>  
> The caller-callerid-number should be of the agent I believe?
>  
> Should I add something like fifo_orbit_exten or fifo_consumer_id ?
>  
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gabe at gundy.org  Tue Sep 25 00:48:59 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 24 Sep 2012 14:48:59 -0600
Subject: [Freeswitch-users] Newbie question,
	FS cli and db (users for user directory)
In-Reply-To: <CAMwZ-tLXr-a0sGCC7DiM5S42ideZPyZP-vtBTueswBWL53D_sA@mail.gmail.com>
References: <CACSMF4yrjxh2h1xNNnFpMr4V_1RvTPJchKgf-Z6fZPqmCDo_Tg@mail.gmail.com>
	<CAKzWOxWYmQiaeDwJpF3ORUZRYpr4GOQAzAFWX19zf5LV+FBa5g@mail.gmail.com>
	<C91105A7-4F54-4D15-A0C7-91667E795C58@gmail.com>
	<CAMwZ-tK64m5CshQpE0JoM+YuCBC8wUtTwYqFom06QLGBA+=10Q@mail.gmail.com>
	<C43E3862-427B-454D-AA1C-F59AB3F0F391@gmail.com>
	<CAMwZ-tLJjf02vACgORBebergpQssftUOU=WT=69dc_DeiunOJQ@mail.gmail.com>
	<CAMwZ-tLXr-a0sGCC7DiM5S42ideZPyZP-vtBTueswBWL53D_sA@mail.gmail.com>
Message-ID: <CAMwZ-tLxP9WzyYJaT75AZ2He+71p+OdOqNQq360CUP4+Z_MEPg@mail.gmail.com>

On Mon, Sep 24, 2012 at 3:19 AM, Gabriel Gunderson <gabe at gundy.org> wrote:
>>> "Another tip is to use cacheable='true' attribute of user xml returned by
>>> web service. This will make FS to maintain a local cache of the account
>>> details. The details will stay in cache until cleared. This means that FS
>>> won't be hitting your web app for every REGISTER request that hoards of SIP
>>> UAs are sending, it will ask only once upon first request. The subsequent
>>> requests will be looked up from cache."
>>
>> Ok, I see what you're talking about. I think what you're looking at
>> isn't mod_xml_curl wide. It seems to be specific to user registration.
>> I'm not sure about this, but I don't see the word 'cacheable' in the
>> mod_xml_curl source, but you do see it on line 1946 of switch_xml.c.
>> And there too, it appears to have something to do with users.
>
> Great... now you've got me thinking about this. I'd like to satisfy my
> curiosity, but naturally, there's nothing in the wiki and no comments
> in the source code :)
>
> I wonder, what does it mean to be 'cached'? And when does that cache
> get cleared? Now I must know ;)

UPDATE: I got a chance to chat with anthm on IRC about this. I'm going
up update the wiki, but only after I play with it for a bit. I'm
posting the chat (cleaned up) here for archival purposes .

******************************************************************************
<gundy> anthm, I try not to bug you as I know you're busy... but I
don't know who else would be involved with this Q.
<gundy> anthm, do you know who might know about an undocumented
"cacheable" parameter when returning directories via mod_xml_curl?
<gundy> anthm, If I can get some clarity on it, I'll update the wiki.

<anthm> gundy, sure

<gundy> anthm, specifically, this is not a mod_xml_curl feature.
Meaning, you can't expect to do that "cacheable" in other responses.
<gundy> anthm, and I guess their isn't any info about when that cache
is cleared.

<anthm> if xml lookup returns a <user> tag with the param
cacheable="true" , it saves it into a in-memory hash table and will
pick it up in place of doing another lookup for the same user later

<gundy> anthm, super, and how long does that cache last and is there a
way to clear it?

<anthm> xml_flush_cache api call
<anthm> xml_flush_cache <id> <key> <val>
<anthm> the less specific you are the more it clears

<gundy> Awesome.

<anthm> so no args is complete clear

<gundy> You have my word, I'll doc it ;) This is only for users, right?

<anthm> yes

<gundy> anthm, thank you for your time.
******************************************************************************

Best,
Gabe


From gabe at gundy.org  Tue Sep 25 01:09:18 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 24 Sep 2012 15:09:18 -0600
Subject: [Freeswitch-users] Apache
In-Reply-To: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>
References: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>
Message-ID: <CAMwZ-tLXR42ybz-8iG3YNMSf4sy3MjOByhZ0pquwWXRbr-+Cug@mail.gmail.com>

On Mon, Sep 24, 2012 at 1:17 PM,  <onyeagbaikenna04 at gmail.com> wrote:
> How can I get freeswitch running on apache 2..
> Thanks for your anticipated response

Start here:
http://www.catb.org/esr/faqs/smart-questions.html#intro

And we'll see where it goes from there.


Best,
Gabe


From bdfoster at endigotech.com  Tue Sep 25 01:12:11 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 24 Sep 2012 17:12:11 -0400
Subject: [Freeswitch-users] voicebox email notifications on Debian with
	exim4
In-Reply-To: <50604618.4010602@gmail.com>
References: <50604618.4010602@gmail.com>
Message-ID: <CAKV-jFzQdV5biKpWagccJ0CjO5hXUaSMAD77sa_LZC1KRH948A@mail.gmail.com>

Im pretty sure this is mentioned already on the wiki, but please elaborate
your process to fill in some gaps.

-BDF
On Sep 24, 2012 8:33 AM, "Daniel-Constantin Mierla" <miconda at gmail.com>
wrote:

> Hello,
>
> I got into the crashing issue for freeswitch voicemail email
> notifications on Debian with exim4, reported at:
>    * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>
> My solution was to use msmtp alongside exim4 -- msmtp is used to relay
> the emails to the local exim4 instance. All seems to work ok, I wonder
> if anyone else is using same approach and got any issues or there are
> other solutions found meanwhile for debian+exim4 combination.
>
> I plan to put more details there, once the registration process for my
> account on the wiki is completed.
>
> 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
> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -
> http://asipto.com/u/katu
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Sep 25 02:14:03 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Mon, 24 Sep 2012 23:14:03 +0100
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <20120924135511.7164ebb726ac4266591be1ee@Vex.Net>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<C067307A-A078-44DE-A922-37F436C3AA36@insensate.co.uk>
	<20120924135511.7164ebb726ac4266591be1ee@Vex.Net>
Message-ID: <4CD0B97E-94B4-49D4-956D-35F8E32DC672@insensate.co.uk>

Hi again,
 Interesting -- socket error crasher. OK.
This stuff -should- be pretty simple. Normally, I write my own (or get some poor programmer to suffer [for] me), so I haven't looked at mod_enum's code yet.

Q: What version of FS were you using (and what platform)?
I'll set up a test config and see if I can reproduce this (won't be until the weekend, I'm afraid).

all the best,
  Lawrence

On 24 Sep 2012, at 18:55, D'Arcy Cain wrote:
> On Sun, 23 Sep 2012 22:54:53 +0100
> Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>> Hi there,
>> full disclosure -- on ENUM, I'm biased :).
> 
> Not having much joy with enum.  I tried the CLI example and got nothing
> back.  I also tried some other numbers including ones suggested on the
> e164.org site with no better luck.  I then tried one of my own numbers
> and FreeSWITCH crashed with a socket error.
> 
> -- 
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.Net



From abaci64 at gmail.com  Tue Sep 25 03:13:22 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 24 Sep 2012 19:13:22 -0400
Subject: [Freeswitch-users] Timezone with time conditions
In-Reply-To: <CALNZuw=wYUrgL5SYttrR9=0U8iFq8ehz0f6ALU5s0Y67hAMBRg@mail.gmail.com>
References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com>
	<CAKzWOxXPWLFCBbM6r0GwjNBW+vqn071k+CUbzBQfx+UJz2ofrQ@mail.gmail.com>
	<CALNZuw=iZq-kq8uxNxpdifW9AdSMbu3LuYBZh9QdeRh5+jW+KQ@mail.gmail.com>
	<CAKzWOxXnuOM2r2sGKi=Hf9CiU+JSzji_=jZxsfVQMOkyb5xA+A@mail.gmail.com>
	<CALNZuw=b9HH=UOemnim-=L0VDsMv3tg9uSRVKHcZae6OuGnHbQ@mail.gmail.com>
	<CAKzWOxWK467=NRc8vAO2iw8TQuzGxjm5zoLC5EN=FD=HCKbaNg@mail.gmail.com>
	<CAKbxfG-A2-4xUEimB-9E971tQWPPx-QtXmyLUXP45UKLrBScAA@mail.gmail.com>
	<CAKbxfG9iW+Jtvdr3KwvZv+fHgZSQ6UH5-CxCb8C9P62Z6qw2_Q@mail.gmail.com>
	<CALNZuw=wYUrgL5SYttrR9=0U8iFq8ehz0f6ALU5s0Y67hAMBRg@mail.gmail.com>
Message-ID: <5060E912.9020005@gmail.com>

I try both "tz-offset" (per tag) and "tod_tz_offset" and it doesn't seem 
to have any effect can someone please confirm that I'm doing the right 
thing and is someone else using it successfully?
Here is my test dialplan:
   <section name="dialplan" description="Regex/XML Dialplan">
     <context name="public">
       <extension name="test">
         <condition break="never">
           <action application="set" data="ringback=$${us-ring}"/>
           <action application="set" data="timezone=America/New_York" 
inline="true"/>
           <action application="set" data="tod_tz_offset=-4" inline="true"/>
           <action application="log" data="CRIT 
${strftime_tz(America/New_York %b %d %Y %H:%M:%S)}"/>
         </condition>
         <condition tz-offset="-4" mon="9" mday="24" 
minute-of-day="900-1200"  break="on-true">
           <action application="transfer" data="30 XML default" />
         </condition>
       </extension>
     </context>
   </section>
I see the log logging the correct time, however the second condition is 
never getting matched based on the tod_tz_offset variable or tz-offset 
tag. if anyone can point me in the right direction I will update the 
documentation.


On 11/30/2011 7:10 AM, Avi Marcus wrote:
> Thanks Anthony!
>
> I update the wiki at http://wiki.freeswitch.org/wiki/Tod if anyone 
> else can test and confirm this feature (on git > 2011-11-08 / revision 
> 65a756643a45a9462270a83238a396d742b7103f)
>
> The TOD doc page could be streamlined a bit, too..
>
> -Avi
>
> On Tue, Nov 8, 2011 at 12:39 AM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     I looked into what the code would require and decided to push a
>     patch, now you get to test and document it.
>
>     if you get the variable tod_tz_offset set on the channel before it
>     hits the dialplan extension in question it will apply that offset
>     to gmt so you can set it to say, -6 for central time.
>
>     Note, you must have it set first so you either need to use inline
>     set or maybe it will work if you have already defined it in your
>     user directory.
>
>     also on a per-tag basis, you can do tz-offset attr ...
>
>
>
>
>     On Mon, Nov 7, 2011 at 4:20 PM, Anthony Minessale
>     <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>
>     wrote:
>
>         on multi home you are better off exploring xml_curl to do time
>         of day stuff, what you suggest would require some new code for
>         sure.
>
>
>         On Mon, Nov 7, 2011 at 4:10 PM, Michael Collins
>         <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>
>             On Mon, Nov 7, 2011 at 11:37 AM, Avi Marcus
>             <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>
>                     Note: The dialplan TOD conditions do not account
>                     for TZ 
>
>                 I think that's the point of this thread. Is there a
>                 point to re-invent the wheel doing TOD conditions
>                 manually?
>
>
>             No, the issue is a matter of supply and demand. The skills
>             needed to add TZ directly into the dialplan are in short
>             supply, whereas doing TZ stuff manually in the dialplan is
>             all but trivial and does not require C coding skills.
>             Also, the demand for TZ checking is relatively low.
>
>             If it's really desired then a bounty should be put up.
>
>             -MC
>
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at 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>
>
>
>
>
>     -- 
>     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>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Sep 25 03:14:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Sep 2012 16:14:17 -0700
Subject: [Freeswitch-users] Freeswitch on Apache
In-Reply-To: <909083812-1348514189-cardhu_decombobulator_blackberry.rim.net-2036827440-@b12.c16.bise7.blackberry>
References: <909083812-1348514189-cardhu_decombobulator_blackberry.rim.net-2036827440-@b12.c16.bise7.blackberry>
Message-ID: <CAKzWOxW82U5ykFZHq0ayymziGChp+SHf8NtrWDrwchT=zUy4Hw@mail.gmail.com>

This counts as an "overly broad" question. Please provide some details.
Also, check out this post
<http://www.catb.org/esr/faqs/smart-questions.html#intro>by Eric S. Raymond
- it describes some techniques you can employ to ask better questions on
community mailing lists. It will save everyone - yourself included - a lot
of time in the long run.

Thanks!
-MC

On Mon, Sep 24, 2012 at 12:18 PM, <onyeagbaikenna04 at gmail.com> wrote:

> Hi everyone
> How can I get freeswitch working on apache 2..
> Thanks for your anticipated response
> Sent from my BlackBerry smartphone from Virgin Media
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 25 03:15:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Sep 2012 16:15:01 -0700
Subject: [Freeswitch-users] Apache
In-Reply-To: <CAMwZ-tLXR42ybz-8iG3YNMSf4sy3MjOByhZ0pquwWXRbr-+Cug@mail.gmail.com>
References: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>
	<CAMwZ-tLXR42ybz-8iG3YNMSf4sy3MjOByhZ0pquwWXRbr-+Cug@mail.gmail.com>
Message-ID: <CAKzWOxWOo3M7Qqhzxo_3HXJu9GSFjz00Y0O=x4yZ2Kio0n9g=g@mail.gmail.com>

Doh, I didn't see this thread when I posted to the previous thread! Well,
at least we're on the same page...
-MC

On Mon, Sep 24, 2012 at 2:09 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Sep 24, 2012 at 1:17 PM,  <onyeagbaikenna04 at gmail.com> wrote:
> > How can I get freeswitch running on apache 2..
> > Thanks for your anticipated response
>
> Start here:
> http://www.catb.org/esr/faqs/smart-questions.html#intro
>
> And we'll see where it goes from there.
>
>
> 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From gabe at gundy.org  Tue Sep 25 03:45:58 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 24 Sep 2012 17:45:58 -0600
Subject: [Freeswitch-users] Apache
In-Reply-To: <CAKzWOxWOo3M7Qqhzxo_3HXJu9GSFjz00Y0O=x4yZ2Kio0n9g=g@mail.gmail.com>
References: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>
	<CAMwZ-tLXR42ybz-8iG3YNMSf4sy3MjOByhZ0pquwWXRbr-+Cug@mail.gmail.com>
	<CAKzWOxWOo3M7Qqhzxo_3HXJu9GSFjz00Y0O=x4yZ2Kio0n9g=g@mail.gmail.com>
Message-ID: <CAMwZ-tJ85zhLzAz7xTBLyTA4CBnszbCKEn-V6YF8u3yFGbV29w@mail.gmail.com>

On Mon, Sep 24, 2012 at 5:15 PM, Michael Collins <msc at freeswitch.org> wrote:
> Doh, I didn't see this thread when I posted to the previous thread! Well, at
> least we're on the same page...

As usual, you handled it with aplomb. Much better than my mild snark.
That's why the FS community needs you!


Gabe


From onyeagbaikenna04 at gmail.com  Tue Sep 25 08:49:40 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Tue, 25 Sep 2012 04:49:40 +0000
Subject: [Freeswitch-users] Apache
In-Reply-To: <CAKzWOxWOo3M7Qqhzxo_3HXJu9GSFjz00Y0O=x4yZ2Kio0n9g=g@mail.gmail.com>
References: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>
	<CAMwZ-tLXR42ybz-8iG3YNMSf4sy3MjOByhZ0pquwWXRbr-+Cug@mail.gmail.com>
	<CAKzWOxWOo3M7Qqhzxo_3HXJu9GSFjz00Y0O=x4yZ2Kio0n9g=g@mail.gmail.com>
Message-ID: <776614911-1348548445-cardhu_decombobulator_blackberry.rim.net-610485308-@b12.c16.bise7.blackberry>

Thanks for all your suggestions, I asked this question in the wrong forum. To all my more experienced colleagues, thanks for the constructive criticism, well noted. 
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: Mon, 24 Sep 2012 16:15:01 
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] Apache

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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 Sep 25 10:17:39 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Tue, 25 Sep 2012 09:17:39 +0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <5060B7C5.1060003@bsd.com.br>
References: <5060B7C5.1060003@bsd.com.br>
Message-ID: <50614C83.1070300@gmail.com>

On 24.09.2012 22:43, Tiago N. Sampaio wrote:
> Hi All!
>
> Im facing a strange issue..
> My recordings (in mp3, wav or gsm) are too fast..
> now recordings are completely inaudible, only with noise and voice
> metallized...
> But sometimes it not happen, just speed issue...
>
> im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2)
>
> today i tryed to change to 1.3.0, but faced another problem, noise on calls.
> i  checked load on my machine, and load was 0.2, almost 100% cpu FREE..
> i have a khomp board with R2 signaling..
>
> Can you help me to debug what is happening?
> Im quite new to FS!!!
>
> Hugs

Please update to the latest developing version using
git checkout
make clean

This happens when it is recorded a transcoded bridge, ie, one leg have 
one codec and another leg - another codec.

Also checkout VoIP phone's settings. If you see "signaling standard" is 
set to "Chinese" change it to some European standard.

-- 
Mimiko desu.


From peetzer at gmail.com  Tue Sep 25 09:46:01 2012
From: peetzer at gmail.com (Peter van Raamsdonk)
Date: Tue, 25 Sep 2012 07:46:01 +0200
Subject: [Freeswitch-users] Mod curl, user directory and Gtalk
Message-ID: <CACSMF4zfxgnko=xwM+T3vOKpAZomu1SSegJZiaQ_7RKJhW5WOw@mail.gmail.com>

Morning list,

I posted my question before but want to elaborate a bit...
For the mod curl my web servlet returns the user directory extensions in
xml,

<document type="freeswitch/xml">
<section name="directory" description="user directory">
<domain name="my.domain">

<!-- add all registered users, user 1001, 1002, 1003 etc -->

<user id="1001">
<params>
<param name="password" value="xxxx"/>
<param name ="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
</params>
<variables>
<variable name="accountcode" value="1001"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Extension 1001"/>
<variable name="effective_caller_id_number" value="1001"/>
<variable name="user_context" value="default"/>
</variables>
</user>

</domain>
</section>
</document>
This works fine and the SIP phones are found by FS.
Is it possible to include also Gtalk profile params in the extension to
register Gtalk extensions?

http://wiki.freeswitch.org/wiki/Dingaling#Sample_Configuration_.28Google.29

Kind regards,

Peter
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From gabe at gundy.org  Tue Sep 25 10:51:30 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 25 Sep 2012 00:51:30 -0600
Subject: [Freeswitch-users] Apache
In-Reply-To: <776614911-1348548445-cardhu_decombobulator_blackberry.rim.net-610485308-@b12.c16.bise7.blackberry>
References: <1549664331-1348514109-cardhu_decombobulator_blackberry.rim.net-732140986-@b12.c16.bise7.blackberry>
	<CAMwZ-tLXR42ybz-8iG3YNMSf4sy3MjOByhZ0pquwWXRbr-+Cug@mail.gmail.com>
	<CAKzWOxWOo3M7Qqhzxo_3HXJu9GSFjz00Y0O=x4yZ2Kio0n9g=g@mail.gmail.com>
	<776614911-1348548445-cardhu_decombobulator_blackberry.rim.net-610485308-@b12.c16.bise7.blackberry>
Message-ID: <CAMwZ-tJsgMnHzjZrr8uJVny5FLjmX474r4YKEOrprNAqBiCt5Q@mail.gmail.com>

On Mon, Sep 24, 2012 at 10:49 PM,  <onyeagbaikenna04 at gmail.com> wrote:
> Thanks for all your suggestions, I asked this question in the wrong forum. To all my more experienced colleagues, thanks for the constructive criticism, well noted.

You're a good sport! Happy hacking!

Gabe


From mike at jerris.com  Tue Sep 25 15:58:04 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Sep 2012 07:58:04 -0400
Subject: [Freeswitch-users] xml tags
In-Reply-To: <39810AE90DFB4F3399336768F94B9CB9@gmail.com>
References: <39810AE90DFB4F3399336768F94B9CB9@gmail.com>
Message-ID: <-4956768462570948262@unknownmsgid>

Typically we use _ for variables and - for params in config.  - is still
valid in json, you just need to reference it differently.  Why would you do
anything to modify them?

On Sep 11, 2012, at 12:30 PM, Seven Du <dujinfang at gmail.com> wrote:

 Hi,

in conference xml_list some tags has "_" but some use "-"

"_" works well but "-" causing problems e.g. when convert to json it's ok
to use conference.last_talking but to get the input-volume value it has to
be conference["input-volume"], and in some other cased there's no such work
around and I have to go through all tags and replace "-"s to "_"s.

I understand "-" in xml is totally valid but would it be better to keep it
consistent ?

        <last_talking>0</last_talking>
        <energy>300</energy>
        <volume_in>0</volume_in>
        <volume_out>0</volume_out>
        <output-volume>0</output-volume>
        <input-volume>0</input-volume>
        <auto-adjusted-input-volume>0</auto-adjusted-input-volume>



Thanks.

-- 
Seven Du
Sent with Sparrow <http://www.sparrowmailapp.com/?sig>

_________________________________________________________________________
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consulting at freeswitch.org
http://www.freeswitchsolutions.com




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From Alexander.Haugg at c4b.de  Tue Sep 25 12:54:26 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Tue, 25 Sep 2012 08:54:26 +0000
Subject: [Freeswitch-users] Consultation Call via event_socket interface
In-Reply-To: <CAKzWOxVHmM5vSZvYBvc21qgscMJJ5eWERTh1v+JV0cyuTrK93Q@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC801E6D7@vc4bex2.corp.c4b.de>
	<CAKzWOxX1oJbzYDSB7GCAi3ZwwNfGrr5jLMfgc-k43ESJRsVsbQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC801EA58@vc4bex2.corp.c4b.de>
	<CABH8QX2ag=R5TPT8nGieuz8ddmLVO46ERWjnKKhvgS7taJh20Q@mail.gmail.com>
	<CAKzWOxWFgXz5fnDUStz4q53x0C3vRdrR4LFPsY2c9u6f69a2AA@mail.gmail.com>
	<17EE12D5-CC1D-49B4-829B-69EC21111547@freeswitch.org>
	<CAKzWOxUEn9-46qQk2F3-=GonkUQrxfZThZ1-JDrSu-DiZjsicw@mail.gmail.com>
	<20E48A825239435BB030CC5C30565542@gmail.com>
	<CAKzWOxVHmM5vSZvYBvc21qgscMJJ5eWERTh1v+JV0cyuTrK93Q@mail.gmail.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC807829F@vc4bex2.corp.c4b.de>

Hi all

The principle control for a consultation call via uuid commands works fine (See my mail from On Sep 17, 2012 3:53 AM)

Now if have a new problem. The channel variable "park_after_bridge" don't work in all cases.

Good scenario:
If i have the legs A (999849368922) and B (170) bridged and i was set for the two legs with uuid_setvar the value park_after_bridge to true.

Now i Hangup the telefone 170 and the leg A go back to Park. Thats correct
2012-09-25 09:47:52.928287 [NOTICE] sofia.c:711 Hangup sofia/external/170 at 172.16.1.26 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2012-09-25 09:47:52.928287 [DEBUG] switch_channel.c:2905 Send signal sofia/external/170 at 172.16.1.26 [KILL]
2012-09-25 09:47:52.928287 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/170 at 172.16.1.26 [BREAK]
2012-09-25 09:47:52.928287 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/170 at 172.16.1.26]
2012-09-25 09:47:52.928287 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/999849368922 at 172.16.1.26 [BREAK]
2012-09-25 09:47:52.928287 [DEBUG] switch_core_state_machine.c:443 (sofia/external/170 at 172.16.1.26) State EXCHANGE_MEDIA going to sleep
2012-09-25 09:47:52.928287 [DEBUG] switch_core_state_machine.c:385 (sofia/external/170 at 172.16.1.26) Running State Change CS_HANGUP
2012-09-25 09:47:52.928287 [DEBUG] switch_core_state_machine.c:625 (sofia/external/170 at 172.16.1.26) State HANGUP
2012-09-25 09:47:52.928287 [DEBUG] switch_ivr_bridge.c:501 sofia/external/170 at 172.16.1.26 ending bridge by request from write function
2012-09-25 09:47:52.928287 [DEBUG] mod_sofia.c:469 Channel sofia/external/170 at 172.16.1.26 hanging up, cause: NORMAL_CLEARING
2012-09-25 09:47:52.928287 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/999849368922 at 172.16.1.26]
2012-09-25 09:47:52.928287 [DEBUG] switch_core_state_machine.c:47 sofia/external/170 at 172.16.1.26 Standard HANGUP, cause: NORMAL_CLEARING
2012-09-25 09:47:52.928287 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/170 at 172.16.1.26 [BREAK]
2012-09-25 09:47:52.928287 [DEBUG] switch_core_state_machine.c:625 (sofia/external/170 at 172.16.1.26) State HANGUP going to sleep
2012-09-25 09:47:52.928287 [DEBUG] switch_core_state_machine.c:416 (sofia/external/170 at 172.16.1.26) State Change CS_HANGUP -> CS_REPORTING
2012-09-25 09:47:52.928287 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/170 at 172.16.1.26 [BREAK]
2012-09-25 09:47:52.928287 [DEBUG] switch_core_state_machine.c:385 (sofia/external/170 at 172.16.1.26) Running State Change CS_REPORTING
2012-09-25 09:47:52.943911 [DEBUG] switch_ivr_bridge.c:1456 sofia/external/170 at 172.16.1.26 skip receive message [UNBRIDGE] (channel is hungup already)
2012-09-25 09:47:52.943911 [DEBUG] switch_core_state_machine.c:685 (sofia/external/170 at 172.16.1.26) State REPORTING
2012-09-25 09:47:52.943911 [DEBUG] switch_core_session.c:773 Send signal sofia/external/999849368922 at 172.16.1.26 [BREAK]
2012-09-25 09:47:52.943911 [DEBUG] switch_ivr.c:2732 (sofia/external/999849368922 at 172.16.1.26) State Change CS_SOFT_EXECUTE -> CS_PARK
2012-09-25 09:47:52.943911 [DEBUG] switch_core_state_machine.c:79 sofia/external/170 at 172.16.1.26 Standard REPORTING, cause: NORMAL_CLEARING

Bad scenario:
If i have the legs A (999849368922) and B (202) bridged i set for the two legs with uuid_setvar the value park_after_bridge to true.

Now i Hangup the telefone 202 and the Freeswitch send a By fort he leg A

2012-09-25 10:24:02.916087 [NOTICE] sofia.c:711 Hangup sofia/external/202 at 172.16.1.26 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2012-09-25 10:24:02.947336 [DEBUG] switch_channel.c:2905 Send signal sofia/external/202 at 172.16.1.26 [KILL]
2012-09-25 10:24:02.947336 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/202 at 172.16.1.26 [BREAK]
2012-09-25 10:24:02.947336 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/202 at 172.16.1.26]
2012-09-25 10:24:02.947336 [DEBUG] switch_channel.c:2882 (sofia/external/999849368922 at 172.16.1.26) Callstate Change ACTIVE -> HANGUP
2012-09-25 10:24:02.947336 [NOTICE] switch_ivr_bridge.c:601 Hangup sofia/external/999849368922 at 172.16.1.26 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2012-09-25 10:24:02.947336 [DEBUG] switch_channel.c:2905 Send signal sofia/external/999849368922 at 172.16.1.26 [KILL]
2012-09-25 10:24:02.947336 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/999849368922 at 172.16.1.26 [BREAK]
2012-09-25 10:24:02.947336 [DEBUG] switch_channel.c:1741 (sofia/external/202 at 172.16.1.26) Callstate Change HANGUP -> ACTIVE
2012-09-25 10:24:02.947336 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/999849368922 at 172.16.1.26 [BREAK]
2012-09-25 10:24:02.947336 [DEBUG] switch_ivr_bridge.c:1456 sofia/external/999849368922 at 172.16.1.26 skip receive message [UNBRIDGE] (channel is hungup already)
2012-09-25 10:24:02.947336 [DEBUG] switch_ivr_bridge.c:1459 sofia/external/202 at 172.16.1.26 skip receive message [UNBRIDGE] (channel is hungup already)
2012-09-25 10:24:02.947336 [DEBUG] switch_core_state_machine.c:446 (sofia/external/202 at 172.16.1.26) State SOFT_EXECUTE going to sleep
2012-09-25 10:24:02.947336 [DEBUG] switch_core_state_machine.c:385 (sofia/external/202 at 172.16.1.26) Running State Change CS_HANGUP
2012-09-25 10:24:02.962960 [DEBUG] switch_ivr_play_say.c:1682 done playing file local_stream://moh
2012-09-25 10:24:02.962960 [DEBUG] switch_core_session.c:2329 sofia/external/999849368922 at 172.16.1.26 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-09-25 10:24:02.962960 [DEBUG] switch_ivr_bridge.c:329 sofia/external/999849368922 at 172.16.1.26 skip receive message [BRIDGE] (channel is hungup already)
2012-09-25 10:24:02.962960 [DEBUG] switch_ivr_bridge.c:331 Send signal sofia/external/202 at 172.16.1.26 [BREAK]
2012-09-25 10:24:02.962960 [DEBUG] switch_ivr_bridge.c:507 sofia/external/999849368922 at 172.16.1.26 ending bridge by request from read function
2012-09-25 10:24:02.962960 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/999849368922 at 172.16.1.26]
2012-09-25 10:24:02.962960 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/202 at 172.16.1.26 [BREAK]
2012-09-25 10:24:02.962960 [DEBUG] switch_core_state_machine.c:443 (sofia/external/999849368922 at 172.16.1.26) State EXCHANGE_MEDIA going to sleep

The only difference is the 202 is an IP-Telephone (not SIP) and the 170 is a System-Telephone.

Have you any idea what is going on in the wrong scenario??

Thak you for your help

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins
Gesendet: Dienstag, 18. September 2012 19:38
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Consultation Call via event_socket interface

Seven,

I think you're right about att_xfer being a tool for a different job. I'm also interested in knowing if using a conference with the mute/unmute and relations is a feasible approach. However, I think the OP may be able to do this all with a little bit of scripting and maybe uuid_dual_transfer + uuid_bridge.

I'll ruminate on this one a bit and maybe we can kick it around after the featured presentation on tomorrow's conference call.

-MC
On Mon, Sep 17, 2012 at 4:49 PM, Seven Du <dujinfang at gmail.com<mailto:dujinfang at gmail.com>> wrote:
I had the similar issue, when using mod_fifo with mixed inbound/outbound agents, I have to disable the multiline feature on hard phones (any one knows how to disable on eye beam or xlite?)

att_xfer could do this job except it has problems when use with event_socket http://jira.freeswitch.org/browse/FS-4419
, maybe it's not designed to this kind of usage.

Yes, I would use the similar approach with Alexander if I'm doing this. Also I wonder if it's possible to just use a conference with mute/unmute hear/nohear and relate/unrelate combinations.


--
Seven Du
Sent with Sparrow<http://www.sparrowmailapp.com/?sig>


On Tuesday, September 18, 2012 at 7:19 AM, Michael Collins wrote:
JM,

That actually explains a lot. I appreciate the information. I can definitely see where the relative expense of a hard phone would make this kind of application quite valuable. I hope you guys are able to iron out the details. Also, maybe you could see if Mitch Capper could add line appearances to FSClient. :)

-MC
On Mon, Sep 17, 2012 at 3:42 PM, Jmesquita at freeswitch.org<mailto:Jmesquita at freeswitch.org> <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:

MC, I believe I have to make a statement here in favor of some realities that are different from the ones most people on this list live in. I live in South America and an IP Phone here does not get lower than 120usd per unit. And I am talking about the cheapest yea link phone model. In brazil, where I come from, this is even higher.

Taking that into consideration, several lower end ip phones started to appear as well as hybrid systems where the majority of the extensions are still analog. In a system like that, CTI applications like the one our friend is describing is really the way out to really add value to a solution.

Even on the asterisk world we see applications like these. For example the flash operator panel and the HUD for asterisk. I believe most of us have heard of it before.

Anyhow, I didn't mean to write an essay on the subject but I see this kind of feature being constantly rejected by the community and I really can't understand why that is.

Jo?o Mesquita

On 17/09/2012, at 01:39 p.m., Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:
Or get a hard phone that has a hold button and at least two line keys.
-MC
On Mon, Sep 17, 2012 at 5:19 AM, Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:


>From personal experience, I believe that how you described is the right way to do it. The only caveat is that you will have to add variables do the channels so you can properly track what is going on in the cdrs . If you don't process cdrs, then it is all good. Careful with pickup and such too...
On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:


Hi MC,



thank you for the answer.

To your question, all call legs in this scenario (outgoing or incoming direction) are connected over a sip trunk of a pbx.



Scenario:

My CTI

  |  A

  |  |       event socket

  V  |

Freeswitch         Sip Trunk

<-                           PBX

->



My Client control (CTI) the call legs over the event socket interface and the call legs are only legs over the sip trunk to or from the PBX.



After some tests i have found a possible solution:

-          Leg A and Leg B are bridged (all legs get the flag park_after_bridge = true)

-          For Consultation i park Leg B and transfer this Leg to Moh in my default context of my dialplan

-          i originate Leg C and bridge this Leg with Leg A (Leg C have the flag park_after_bridge = true too) Consultation is comlete now.

-          with the principle same think i can toggle Leg A <-> Leg B and Leg A <-> Leg C.



What is your think for this solution? I have tested this on the FS CLI and it works.



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 Michael Collins
Gesendet: Freitag, 14. September 2012 19:13
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Consultation Call via event_socket interface



Hi Alex,

Welcome to the FreeSWITCH mail list!

First question for you: what kind of telephone are you using? The reason I ask is that this kind of function is trivially achieved with a good hard phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys. If you can use a hard phone w/ multiple line keys then you don't even need to mess with the dialplan, uuid_bridge, etc.

-MC

On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:

Hi All,



I'm new on the mailing list.

I have a problem with a call scenario.

-          Channel A and channel B are bridged (A is my own channel and B is my calling partner)

-          Now i set channel B on hold with the command "uuid_hold xxx" and create a new channel to C with the command:

bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip',origination_caller_id_number=num at ip}sofia/external/num at ip  &park

this works correctly, the partner C answer and the channel is established.

-          Now the Problem:

I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false.

Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment.

Is there a general problem in my plan to do that?

Is it a better plan to do this over the dialplan?

The next step in this scenario is to toggle the connection A -> B and A -> C.



Thanks for your help!

Nice regards,

Alex

_________________________________________________________________________
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--
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>
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_________________________________________________________________________
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--
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
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_________________________________________________________________________
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--
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
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FreeSWITCH-users mailing list
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
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--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From darcy at Vex.Net  Tue Sep 25 16:44:58 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Tue, 25 Sep 2012 08:44:58 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
	<1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>
	<CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>
Message-ID: <20120925084458.697587323d4c362652ca6bc3@Vex.Net>

On Mon, 24 Sep 2012 00:40:37 +0200
Avi Marcus <avi at avimarcus.net> wrote:
> enum or a DB query to your routing table for inbound calls, e.g. using
> mod_odbc_query <http://wiki.freeswitch.org/wiki/Mod_odbc_query>.

After some investigation I stumbled across "user_exists" and thought
that something like this might work in dialplan/default.xml and
public.xml.

 <extension name="user_found">
  <condition field="${user_exists(id ${destination_number} $${domain})}" expression="true"> 
  <action application="transfer" data="${destination_number}" />
</extension>

I also tried "bridge" instead of "transfer" but either way FreeSWITCH
fails to restart.

Stopping FreeSWITCH.
Killing: 14561
Waiting for PIDS: 14561.
Starting freeswitch.
WARNING: Wasting up to 8 megs of memory per thread.
16555 Backgrounding.
FreeSWITCH[16554] System Ready pid:16555
[ERROR] fs_cli.c:1376 main() Error Connecting [Socket Connection Error]

There is nothing in the log.  Can anyone see what I am doing wrong here?

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From lloyd.aloysius at gmail.com  Tue Sep 25 19:14:46 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Tue, 25 Sep 2012 11:14:46 -0400
Subject: [Freeswitch-users] xml_curl not fetching any information from the
	web server
Message-ID: <CAGvuWYbwoyKaGcfg0qUi7xoXJyNYXrGCQAz2hMELb2rQ6YAABA@mail.gmail.com>

Hello,

FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at
commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z

There is no error. I check several times the xml_curl output is right.
There is no error.

switch not getting any information from the web server.


2012-09-25 11:08:25.972070 [NOTICE] mod_xml_curl.c:538 Binding
[configuration] XML Fetch Function* [http://A.B.C.D/xml_curl/index.php]
[configuration]*
2012-09-25 11:08:25.972118 [NOTICE] mod_xml_curl.c:538 Binding [directory]
XML Fetch Function *[http:// A.B.C.D /xml_curl/index.php]
[dialplan|directory]*
2012-09-25 11:08:25.972890 [CONSOLE] switch_loadable_module.c:1328
Successfully Loaded [mod_xml_curl]
2012-09-25 11:08:25.972922 [NOTICE] switch_loadable_module.c:298 Adding API
Function 'xml_curl'


How to trouble shoot this issue. Any help is appreciated.

Than you
Lloyd
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From vipkilla at gmail.com  Tue Sep 25 19:40:33 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Tue, 25 Sep 2012 11:40:33 -0400
Subject: [Freeswitch-users] xml_curl not fetching any information from
 the web server
In-Reply-To: <CAGvuWYbwoyKaGcfg0qUi7xoXJyNYXrGCQAz2hMELb2rQ6YAABA@mail.gmail.com>
References: <CAGvuWYbwoyKaGcfg0qUi7xoXJyNYXrGCQAz2hMELb2rQ6YAABA@mail.gmail.com>
Message-ID: <CAC-LwPOFfisT6CyiuG9S3tV5GoBwQwqSWbgPTRk5x3QAcTdXVg@mail.gmail.com>

Check your webserver access or error log while FreeSWITCH tries to
retrieve data from the webserver.

On Tue, Sep 25, 2012 at 11:14 AM, Lloyd Aloysius
<lloyd.aloysius at gmail.com> wrote:
> Hello,
>
> FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at
> commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z
>
> There is no error. I check several times the xml_curl output is right. There
> is no error.
>
> switch not getting any information from the web server.
>
>
> 2012-09-25 11:08:25.972070 [NOTICE] mod_xml_curl.c:538 Binding
> [configuration] XML Fetch Function [http://A.B.C.D/xml_curl/index.php]
> [configuration]
> 2012-09-25 11:08:25.972118 [NOTICE] mod_xml_curl.c:538 Binding [directory]
> XML Fetch Function [http:// A.B.C.D /xml_curl/index.php]
> [dialplan|directory]
> 2012-09-25 11:08:25.972890 [CONSOLE] switch_loadable_module.c:1328
> Successfully Loaded [mod_xml_curl]
> 2012-09-25 11:08:25.972922 [NOTICE] switch_loadable_module.c:298 Adding API
> Function 'xml_curl'
>
>
> How to trouble shoot this issue. Any help is appreciated.
>
> Than you
> 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
>


From lloyd.aloysius at gmail.com  Tue Sep 25 19:51:33 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Tue, 25 Sep 2012 11:51:33 -0400
Subject: [Freeswitch-users] xml_curl not fetching any information from
 the web server
In-Reply-To: <CAC-LwPOFfisT6CyiuG9S3tV5GoBwQwqSWbgPTRk5x3QAcTdXVg@mail.gmail.com>
References: <CAGvuWYbwoyKaGcfg0qUi7xoXJyNYXrGCQAz2hMELb2rQ6YAABA@mail.gmail.com>
	<CAC-LwPOFfisT6CyiuG9S3tV5GoBwQwqSWbgPTRk5x3QAcTdXVg@mail.gmail.com>
Message-ID: <CAGvuWYY4rX39hvH-pLokD4Ht=O1QkQv=3Vu7ZdaDcW+9Y1FV5g@mail.gmail.com>

I check from the web browser the output is perfect. The same url is use by
freeswitch.

This configuration works all the time. Could not find any reason why the
switch not fetching the configuration




On Tue, Sep 25, 2012 at 11:40 AM, Vik Killa <vipkilla at gmail.com> wrote:

> Check your webserver access or error log while FreeSWITCH tries to
> retrieve data from the webserver.
>
> On Tue, Sep 25, 2012 at 11:14 AM, Lloyd Aloysius
> <lloyd.aloysius at gmail.com> wrote:
> > Hello,
> >
> > FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at
> > commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z
> >
> > There is no error. I check several times the xml_curl output is right.
> There
> > is no error.
> >
> > switch not getting any information from the web server.
> >
> >
> > 2012-09-25 11:08:25.972070 [NOTICE] mod_xml_curl.c:538 Binding
> > [configuration] XML Fetch Function [http://A.B.C.D/xml_curl/index.php]
> > [configuration]
> > 2012-09-25 11:08:25.972118 [NOTICE] mod_xml_curl.c:538 Binding
> [directory]
> > XML Fetch Function [http:// A.B.C.D /xml_curl/index.php]
> > [dialplan|directory]
> > 2012-09-25 11:08:25.972890 [CONSOLE] switch_loadable_module.c:1328
> > Successfully Loaded [mod_xml_curl]
> > 2012-09-25 11:08:25.972922 [NOTICE] switch_loadable_module.c:298 Adding
> API
> > Function 'xml_curl'
> >
> >
> > How to trouble shoot this issue. Any help is appreciated.
> >
> > Than you
> > 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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Sep 25 20:14:21 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 26 Sep 2012 00:14:21 +0800
Subject: [Freeswitch-users] xml tags
In-Reply-To: <-4956768462570948262@unknownmsgid>
References: <39810AE90DFB4F3399336768F94B9CB9@gmail.com>
	<-4956768462570948262@unknownmsgid>
Message-ID: <B5CBCB2941F146B4B3496E91C22BEEFA@gmail.com>

Yes, I understand it's valid json, and as in json there's a difference reference but in my special case I'm in erlang and using the erlydtl template and obviously conference.member.volume_in is valid but conference.member.input-volume is not.

Well, it's not a big deal and I can go over all and replace all '-'s to '_'s before rendering to the template.

Thanks. 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, September 25, 2012 at 7:58 PM, Michael Jerris wrote:

> Typically we use _ for variables and - for params in config.  - is still valid in json, you just need to reference it differently.  Why would you do anything to modify them?
> 
> On Sep 11, 2012, at 12:30 PM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> 
> > Hi, 
> > 
> > in conference xml_list some tags has "_" but some use "-"
> > 
> > "_" works well but "-" causing problems e.g. when convert to json it's ok to use conference.last_talking but to get the input-volume value it has to be conference["input-volume"], and in some other cased there's no such work around and I have to go through all tags and replace "-"s to "_"s. 
> > 
> > I understand "-" in xml is totally valid but would it be better to keep it consistent ? 
> > 
> > <last_talking>0</last_talking> <energy>300</energy> <volume_in>0</volume_in> <volume_out>0</volume_out> <output-volume>0</output-volume> <input-volume>0</input-volume> <auto-adjusted-input-volume>0</auto-adjusted-input-volume>
> > 
> > 
> > Thanks.
> > 
> > -- 
> > Seven Du
> > Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
> > 
> > _________________________________________________________________________
> > 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 lloyd.aloysius at gmail.com  Tue Sep 25 21:28:29 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Tue, 25 Sep 2012 13:28:29 -0400
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FDF1CB@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>
	<CAPcNkddX0xXdJsYm+UTwy=xo5Saw5gxk=5XF_rMcJi7yTZ+mzQ@mail.gmail.com>
	<CAKDfjgcMA7e3Z22=3wcLWVyUz-Ygb3oejmRwGqfc=w_n1wiKJQ@mail.gmail.com>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FDF1CB@BLUPRD0711MB413.namprd07.prod.outlook.com>
Message-ID: <CAGvuWYa-87w5qJHWGQnHRfsvQVTr-uWm8YXHF3RLTyXWqf6-Hg@mail.gmail.com>

Hi All:

Does this performance problem still exists in CentOS 6.3?

Thank you in advance.

Thanks
Lloyd
*
*
*  <http://www.sunteltech.ca/blog/>*



On Fri, Sep 14, 2012 at 6:48 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

> I found someone with more info on scheduling problems, especially with
> sleep/wait, of which FS does a lot. There's a parameter called
> sched_migration_cost which might be worth investigating.
>
> http://pastebin.freeswitch.com/19903
>
> On a benchmark for sleeps, he says RHEL6 is taking up to 45% more time.
> With the huge number of threads and sleeps FS does, maybe those little
> differences add up:
> http://unix.stackexchange.com/questions/37391/high-cpu-usage-with-cfs
>
> Separately, to everyone that hasn't rebooted their server in the last few
> months, you may still have the leap-second set which will cause lots of CPU
> spinning due to a kernel bug. Running date -s "`date`" should fix that. I
> am pretty sure that freed up at least one core on one of our boxes that was
> still spinning since Jul 1.
>
> We've now upgraded some systems to CentOS 6.3 (with all post release
> updates) and left others at 6.2. Over the next week we'll see if they seem
> to be much different. At the moment, the general CPU usage seems to be
> similar; we'll have to pound some dialer on it and see what happens.
>
> -Michael
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
> Kielhofner
> Sent: Monday, September 03, 2012 12:42 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] RedHat 6.X Performance
>
> Anymore info on specific traffic patterns that cause this?  I've been
> beating up FS on 6.3 for several days now and haven't seen anything unusual
> yet...
>
> On Thu, Aug 30, 2012 at 11:45 PM, Ken Rice <krice at freeswitch.org> wrote:
> > 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...
> >
> >
>
> --
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tnsampaio at bsd.com.br  Tue Sep 25 21:31:32 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Tue, 25 Sep 2012 14:31:32 -0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <50614C83.1070300@gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
Message-ID: <5061EA74.5040001@bsd.com.br>

Hi Vbvbrj!

     I updated to 1.2.3 but the problems still occurs...
     Im using Xlite and Zoiper as softphone, both with recording issue.
     I try change codecs, all codecs problem still occurs...


Em 25/09/2012 03:17, Vbvbrj escreveu:
> On 24.09.2012 22:43, Tiago N. Sampaio wrote:
>> Hi All!
>>
>> Im facing a strange issue..
>> My recordings (in mp3, wav or gsm) are too fast..
>> now recordings are completely inaudible, only with noise and voice
>> metallized...
>> But sometimes it not happen, just speed issue...
>>
>> im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2)
>>
>> today i tryed to change to 1.3.0, but faced another problem, noise on calls.
>> i  checked load on my machine, and load was 0.2, almost 100% cpu FREE..
>> i have a khomp board with R2 signaling..
>>
>> Can you help me to debug what is happening?
>> Im quite new to FS!!!
>>
>> Hugs
> Please update to the latest developing version using
> git checkout
> make clean
>
> This happens when it is recorded a transcoded bridge, ie, one leg have
> one codec and another leg - another codec.
>
> Also checkout VoIP phone's settings. If you see "signaling standard" is
> set to "Chinese" change it to some European standard.
>



From mgg at giagnocavo.net  Tue Sep 25 21:55:35 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 25 Sep 2012 17:55:35 +0000
Subject: [Freeswitch-users] RedHat 6.X Performance
Message-ID: <q3s61gai01domhf1pj0of9qn.1348595699781@email.android.com>

We haven't noticed it, but we haven't explicitly checked. 

Lloyd Aloysius <lloyd.aloysius at gmail.com> wrote:


From freeswitch-list at puzzled.xs4all.nl  Tue Sep 25 22:07:33 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 25 Sep 2012 20:07:33 +0200
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAGvuWYa-87w5qJHWGQnHRfsvQVTr-uWm8YXHF3RLTyXWqf6-Hg@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>
	<CAKDfjgcMA7e3Z22=3wcLWVyUz-Ygb3oejmRwGqfc=w_n1wiKJQ@mail.gmail.com>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FDF1CB@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<CAGvuWYa-87w5qJHWGQnHRfsvQVTr-uWm8YXHF3RLTyXWqf6-Hg@mail.gmail.com>
Message-ID: <5061F2E5.3080209@puzzled.xs4all.nl>

On 09/25/2012 07:28 PM, Lloyd Aloysius wrote:
> Hi All:
>
> Does this performance problem still exists in CentOS 6.3?

Afaik this issue is also present in 6.3 although some have reported that 
they do not see the issue while others do. I guess it depends on your setup.

Regards,
Patrick



From vbvbrj at gmail.com  Tue Sep 25 22:12:51 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Tue, 25 Sep 2012 21:12:51 +0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <5061EA74.5040001@bsd.com.br>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br>
Message-ID: <5061F423.8020301@gmail.com>

On 25.09.2012 20:31, Tiago N. Sampaio wrote:
>       I updated to 1.2.3 but the problems still occurs...
>       Im using Xlite and Zoiper as softphone, both with recording issue.
>       I try change codecs, all codecs problem still occurs...

The latest version, I mean v.1.3.0. Set both softphones to same default 
codec PCMU for use. And see the logs when bot legs negotiate codecs.

-- 
Mimiko desu.


From sertys at gmail.com  Tue Sep 25 18:19:52 2012
From: sertys at gmail.com (Daniel Ivanov)
Date: Tue, 25 Sep 2012 07:19:52 -0700
Subject: [Freeswitch-users] SRTP and bridge woes
Message-ID: <CAMeZp60BkdjafMZokBTnga=_Zu-OvwoUGW=Wea4GxWAynf6zWA@mail.gmail.com>

I am building an ostn node. I followed a recipe, but got caught up with
learning FS basics. I had this trouble that calls were being signalled, but
no audio was actually put through.
I thought it was nat problems, but connectivity was going fine.
This is my dialplan snippet:
 <extension name="Local_Extension">
      <condition field="destination_number" expression="^(\d{3})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
       <!-- <bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s]
<app> -->
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
<!--        <action application="bind_meta_app" data="2 b s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set"
data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
        <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<!--       <action application="answer"/>  -->
        <action application="info"/>
         <action application="answer"/>
        <action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>
    <!--    <action application="answer"/> -->
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
      </condition>
    </extension>

This works, but you can see i have an answer app right before the bridge.
If i don't answer the call, i don't get to the point where i get Activating
SRTP RECV/SRTP SEND. Yet both channels get answered and RTP goes both ways,
but i have a feeling it's not SRTP and thus is not being decrypted.
I have this in the global context:
 <condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_80)$" 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>

 which should activate SRTP on both legs.

I have pasted in to the bin a debug log with and without the answer and
media sending here :
http://pastebin.freeswitch.org/19937
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From zhenya.malyshev at onecallnow.com  Tue Sep 25 18:25:52 2012
From: zhenya.malyshev at onecallnow.com (Zhenya Malyshev)
Date: Tue, 25 Sep 2012 10:25:52 -0400
Subject: [Freeswitch-users] [ERR] mod_spidermonkey.c:3439 Error loading ODBC,
 followed by ReferenceError: ODBC is not defined
Message-ID: <6E70188AF42BCB46814B642FFEFB94AA4904243335@VMBX131.ihostexchange.net>

Hello everyone,

I am trying to resolve the issue that was discussed a few times in this user group, but by some reason recommended solutions do not work for me.

While calling MySQL using ODBC from javascript I am getting the famous [ERR] mod_spidermonkey.c:3439 Error loading ODBC, followed by ReferenceError: ODBC is not defined
I searched for it and found a perfect explanation by Michael J. and other group participants how to resolve it:

http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006863.html
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc

I installed unixODBC and found <load mod_spidermonkey_odbc /> in spidermonkey.conf.xml,
but I cannot uncomment the languages/mod_spidermonkey_odbc line within the modules.conf file since there is no such line.
I tried to add it, but got the error while re-compiling. While browsing through this user group discussions I sometimes notice inaccuracy in
recommendations which are no essential for experienced users, but misleading for armatures like myself.

I would greatly appreciate if someone could clarify that languages/mod_spidermonkey_odbc in modules.conf statement.
There is only one line referencing spidermonkey and it is languages/mod_spidermonkey.

Thanks,

Eugene
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From darcy at Vex.Net  Tue Sep 25 19:13:08 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Tue, 25 Sep 2012 11:13:08 -0400
Subject: [Freeswitch-users] Keeping local calls local
In-Reply-To: <20120925084458.697587323d4c362652ca6bc3@Vex.Net>
References: <20120923110121.d4b216eaa13d8073eb2a65d0@Vex.Net>
	<1348424257.74917.YahooMailNeo@web39303.mail.mud.yahoo.com>
	<20120923160441.fe473e06925857e1efa7605d@Vex.Net>
	<1348432289.66315.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<1348437604.32066.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAKV-jFw6uZ-mU=eETkf8+E6_cwXsJ7Cse2VS3W+Pu0n28YHO8Q@mail.gmail.com>
	<1261C560-4B39-4389-B947-416870C03A95@insensate.co.uk>
	<CALNZuwkQUAr1tWvd3nHn-sgUayLFryBMmExMQfcjHDOC+pXj0A@mail.gmail.com>
	<20120925084458.697587323d4c362652ca6bc3@Vex.Net>
Message-ID: <20120925111308.8575cc6fabe0abc0e25828c1@Vex.Net>

On Tue, 25 Sep 2012 08:44:58 -0400
D'Arcy Cain <darcy at Vex.Net> wrote:
>  <extension name="user_found">
>   <condition field="${user_exists(id ${destination_number} $${domain})}" expression="true"> 
>   <action application="transfer" data="${destination_number}" />
> </extension>

Never mind.  BCAS error.  Here is what works.

  <extension name="VEX_user_found">
    <condition field="${user_exists(id ${destination_number} $${domain})}" expression="true">
      <action application="bridge" data="user/${destination_number}@$${domain}"/>
    </condition>
  </extension>

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From tnsampaio at bsd.com.br  Tue Sep 25 22:50:30 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Tue, 25 Sep 2012 15:50:30 -0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <5061EA74.5040001@bsd.com.br>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br>
Message-ID: <5061FCF6.2040902@bsd.com.br>

I forgot to mention, when a call come from PSTN it records fine, in mp3 
or wav.
Im using an R2 KHOMP board.

Its driving me crazy hehe!!!

Thx for help!



Em 25/09/2012 14:31, Tiago N. Sampaio escreveu:
> Hi Vbvbrj!
>
>     I updated to 1.2.3 but the problems still occurs...
>     Im using Xlite and Zoiper as softphone, both with recording issue.
>     I try change codecs, all codecs problem still occurs...
>
>
> Em 25/09/2012 03:17, Vbvbrj escreveu:
>> On 24.09.2012 22:43, Tiago N. Sampaio wrote:
>>> Hi All!
>>>
>>> Im facing a strange issue..
>>> My recordings (in mp3, wav or gsm) are too fast..
>>> now recordings are completely inaudible, only with noise and voice
>>> metallized...
>>> But sometimes it not happen, just speed issue...
>>>
>>> im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2)
>>>
>>> today i tryed to change to 1.3.0, but faced another problem, noise 
>>> on calls.
>>> i  checked load on my machine, and load was 0.2, almost 100% cpu FREE..
>>> i have a khomp board with R2 signaling..
>>>
>>> Can you help me to debug what is happening?
>>> Im quite new to FS!!!
>>>
>>> Hugs
>> Please update to the latest developing version using
>> git checkout
>> make clean
>>
>> This happens when it is recorded a transcoded bridge, ie, one leg have
>> one codec and another leg - another codec.
>>
>> Also checkout VoIP phone's settings. If you see "signaling standard" is
>> set to "Chinese" change it to some European standard.
>>
>



From david.villasmil.work at gmail.com  Wed Sep 26 01:52:57 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 25 Sep 2012 23:52:57 +0200
Subject: [Freeswitch-users] CLI Debug request?
Message-ID: <CAFGRPVr6KgMibcit3r2-s0v--HKiO5s6NtomG9V6P6qCj=o4vA@mail.gmail.com>

Hello All,

It would be VERY nice to be able to set a "fsctl loglevel debug" on a
specific "called number", maybe by looking at the cldnum and filtering all
output but the uniqueID?
And by this I mean Sofia as much as ANY log information happening ONLY for
that specific call..

When one has high loads it is extremely difficult to debug a specific
situation...

Maybe this is already possible, but I don't know how?

Thanks!

David
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From roberto at i360tecnologia.com.br  Wed Sep 26 03:25:28 2012
From: roberto at i360tecnologia.com.br (Roberto Linck)
Date: Tue, 25 Sep 2012 20:25:28 -0300
Subject: [Freeswitch-users] Audio Playback Issue
Message-ID: <CAFTa0yd_zm7hCD0ARjFGK6YgsbMJSfC+MhCVttUsMzgmteoSDw@mail.gmail.com>

Hi everybody, I'm trying to reproduce some wav audio files in a one leg
call. The call is originated programmatically by a C module written by
myself. The issue is when the destination is a SIP or GSM, the audio is
reproduced OK, but when the destination is land line, the audio is chunked
and cutted. I already tried many sorts of sample rate - always on wav
format - but I always had the same result. I really had understood that the
issue is related with the type of destination line, but I'm accepting any
suggestion to solve this.
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From chad at apartmentlines.com  Wed Sep 26 05:10:08 2012
From: chad at apartmentlines.com (Chad Phillips)
Date: Tue, 25 Sep 2012 18:10:08 -0700
Subject: [Freeswitch-users] CLI Debug request?
In-Reply-To: <CAFGRPVr6KgMibcit3r2-s0v--HKiO5s6NtomG9V6P6qCj=o4vA@mail.gmail.com>
References: <CAFGRPVr6KgMibcit3r2-s0v--HKiO5s6NtomG9V6P6qCj=o4vA@mail.gmail.com>
Message-ID: <E1650713BEC34DB2965B6C87734BE3C4@gmail.com>

http://sourceforge.net/projects/pcapsipdump/ is great for per call packet captures. 


On Tuesday, September 25, 2012 at 2:52 PM, David Villasmil wrote:

> Hello All,
> 
> It would be VERY nice to be able to set a "fsctl loglevel debug" on a specific "called number", maybe by looking at the cldnum and filtering all output but the uniqueID?  
> And by this I mean Sofia as much as ANY log information happening ONLY for that specific call..
> 
> When one has high loads it is extremely difficult to debug a specific situation...
> 
> Maybe this is already possible, but I don't know how?
> 
> Thanks!
> 
> David 
> _________________________________________________________________________
> 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 tnsampaio at bsd.com.br  Wed Sep 26 05:55:44 2012
From: tnsampaio at bsd.com.br (Tiago Sampaio)
Date: Tue, 25 Sep 2012 22:55:44 -0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <5061F423.8020301@gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
Message-ID: <CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>

I tested with 1.3.0, and the problem still occours...

Testing to better debug the issue, i can reach this:

PSTN -> FS -> user/1000 (record ok!)
user/1000 -> FS -> user/1001 (record ok!)
user/1000 -> FS -> PSTN problem occurs..

I know this is a transcoding problem, but why when i change codec to PCMA
or PCMU nothing changes..
I tryed all codecs, without success...
My pstn is an MFCr2 E1 circuit over a Khomp board!
Im very lost and confused, this problem is killing me :/



2012/9/25 Vbvbrj <vbvbrj at gmail.com>

> On 25.09.2012 20:31, Tiago N. Sampaio wrote:
> >       I updated to 1.2.3 but the problems still occurs...
> >       Im using Xlite and Zoiper as softphone, both with recording issue.
> >       I try change codecs, all codecs problem still occurs...
>
> The latest version, I mean v.1.3.0. Set both softphones to same default
> codec PCMU for use. And see the logs when bot legs negotiate codecs.
>
> --
> 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
>



-- 
Tiago N. Sampaio
BSD Certified Associate
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From cmrienzo at gmail.com  Wed Sep 26 06:11:57 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 25 Sep 2012 22:11:57 -0400
Subject: [Freeswitch-users] CLI Debug request?
In-Reply-To: <CAFGRPVr6KgMibcit3r2-s0v--HKiO5s6NtomG9V6P6qCj=o4vA@mail.gmail.com>
References: <CAFGRPVr6KgMibcit3r2-s0v--HKiO5s6NtomG9V6P6qCj=o4vA@mail.gmail.com>
Message-ID: <CAD-qRkiz6AA6G3qqGWHx=PodFfN4OyqJHu7pBHPdWtJ4mgZazQ@mail.gmail.com>

There is a session_loglevel application you can call from the dialplan.

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



On Tue, Sep 25, 2012 at 5:52 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello All,
>
> It would be VERY nice to be able to set a "fsctl loglevel debug" on a
> specific "called number", maybe by looking at the cldnum and filtering all
> output but the uniqueID?
> And by this I mean Sofia as much as ANY log information happening ONLY for
> that specific call..
>
> When one has high loads it is extremely difficult to debug a specific
> situation...
>
> Maybe this is already possible, but I don't know how?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From don.dawson at voice-ring.com  Wed Sep 26 06:15:38 2012
From: don.dawson at voice-ring.com (Don Dawson)
Date: Tue, 25 Sep 2012 21:15:38 -0500
Subject: [Freeswitch-users] force 503 or 302 response on REGISTER
Message-ID: <5062654A.3020906@voice-ring.com>


Is there anyway to force a '503' response to a REGISTER sip msg?
Right now, it responds with '404' on a 'not found' curl http xml response.




From valery.kalinin at gmail.com  Wed Sep 26 06:15:11 2012
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Wed, 26 Sep 2012 08:15:11 +0600
Subject: [Freeswitch-users] Empty wiki and my head
Message-ID: <CAF=swh_LmBERMqTx4goR4aoGrKfk4ZzMTAh-CoYmGWS90N7Uog@mail.gmail.com>

I cannot find any documentation for these modules:
mod_abstraction
mod_sonar
mod_html5
mod_xml_scgi
What they do and how to use them?


From the_spide21 at yahoo.com  Wed Sep 26 06:39:44 2012
From: the_spide21 at yahoo.com (Carlos Lopez)
Date: Tue, 25 Sep 2012 19:39:44 -0700 (PDT)
Subject: [Freeswitch-users] Automatic Callout and TTS
In-Reply-To: <1348499545.93808.YahooMailNeo@web45107.mail.sp1.yahoo.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<EDF753B814C44E44BD108AE6DCEF73E8@e1705>
	<1348499545.93808.YahooMailNeo@web45107.mail.sp1.yahoo.com>
Message-ID: <1348627184.87639.YahooMailNeo@web45111.mail.sp1.yahoo.com>

Thank you Hector Geraldino for your guide.

Mainly I use PHP so I'll try to setup a service like app with ESL and PHP support and capture the events so FS can do the callout + TTS.

Carlos.

----------------Wrote-----------

You will need a man-in-the-middle application (broker) that receives the message and converts it to actionable steps that will be performed in FS.

Of the many ways you have to do this, I can suggest you to explore the Event Socket Outbound ? and particulary the Java ESL Client, although you can use pretty much any other language supported by FS. Main idea is: your application will receive the message, and using ESL will generate an outbound call using FreeSWITCH. When the call is answered it will play a message using a TTS engine (flite, cepstral, nuance, etc.), capture the input and do whatever you need with it.

The control of the call can be handled via ESL or using a dialplan.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos Lopez
Sent: Monday, September 24, 2012 11:12 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Automatic Callout and TTS Hi all, I am just learning how to install and use Freeswitch and at the place where I'm working i'd like to implement a solution like callout and TTS. The main goal is to use Freeswith as a PBX and use text based strings so the FSW can callout some customers and advise them about theit current balance so the string will be componded by a set of fields and the resulting text most be converted to phone sound to the customer, ex. 1- Event occurs on our main system server.
2- Server send a sms/E-Mail/HTTP-String to FSW, with customer phone # and string ( String will be like: "You have % credit balance on your account. Please type 1 to add credit", where % is a numeric value that will be filled up by our main system servers.
3- FSW will convert resulting text to sound call and calls the customer's phone number.
4- Customer picks up the phone and listen to the TTS sound. Is it possible with FSW? Thank you all for your response. Carlos.
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From b2m at a-cti.com  Wed Sep 26 10:23:47 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Wed, 26 Sep 2012 11:53:47 +0530
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
Message-ID: <CAP2fbbcY3xE6NTg5CfCHiFgEBdY0TydwXXS6Xsk7RoAWa82EdQ@mail.gmail.com>

I had this issue before, I tried so many things finally this version helped
me. Everything was fine before and some combination of HW/SW did the same
to me.

FreeSWITCH Version 1.0.head (git-9f8d37d 2012-01-28 18-32-35 -0600)

May be this is totally irrelevant to your issue, thought to share my
experience.

Thanks,
Bala

On Wed, Sep 26, 2012 at 7:25 AM, Tiago Sampaio <tnsampaio at bsd.com.br> wrote:

> I tested with 1.3.0, and the problem still occours...
>
> Testing to better debug the issue, i can reach this:
>
> PSTN -> FS -> user/1000 (record ok!)
> user/1000 -> FS -> user/1001 (record ok!)
> user/1000 -> FS -> PSTN problem occurs..
>
> I know this is a transcoding problem, but why when i change codec to PCMA
> or PCMU nothing changes..
> I tryed all codecs, without success...
> My pstn is an MFCr2 E1 circuit over a Khomp board!
> Im very lost and confused, this problem is killing me :/
>
>
>
> 2012/9/25 Vbvbrj <vbvbrj at gmail.com>
>
>> On 25.09.2012 20:31, Tiago N. Sampaio wrote:
>> >       I updated to 1.2.3 but the problems still occurs...
>> >       Im using Xlite and Zoiper as softphone, both with recording issue.
>> >       I try change codecs, all codecs problem still occurs...
>>
>> The latest version, I mean v.1.3.0. Set both softphones to same default
>> codec PCMU for use. And see the logs when bot legs negotiate codecs.
>>
>> --
>> 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
>>
>
>
>
> --
> Tiago N. Sampaio
> BSD Certified Associate
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Sep 26 11:05:14 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 26 Sep 2012 10:05:14 +0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
Message-ID: <5062A92A.2090808@gmail.com>

On 26.09.2012 04:55, Tiago Sampaio wrote:
> I tested with 1.3.0, and the problem still occours...
>
> Testing to better debug the issue, i can reach this:
>
> PSTN -> FS -> user/1000 (record ok!)
> user/1000 -> FS -> user/1001 (record ok!)
> user/1000 -> FS -> PSTN problem occurs..

So problem occur when an VoIP client calls the PSTN client, ie an 
outbound call from FS to PSTN. Checkout in vars.xml the option 
outbound_codec_prefs and it must contain the same codecs, especialy PCMU 
or PCMA.

-- 
Mimiko desu.


From peter.olsson at visionutveckling.se  Wed Sep 26 12:13:20 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 26 Sep 2012 08:13:20 +0000
Subject: [Freeswitch-users] [ERR] mod_spidermonkey.c:3439 Error loading
 ODBC, followed by ReferenceError: ODBC is not defined
Message-ID: <1FFF97C269757C458224B7C895F35F1515D7A1@cantor.std.visionutv.se>

mod_spidermonkey_odbc is not a FreeSWITCH module, it's a spidermonkey submodule, so there is no need to put it in modules.conf. If you have ODBC installed when you run ./bootstrap.sh and ./configure, it will autodetect it, and build the mod_spidermonkey_odbc module. The only thing you need to do after that is to add it to spidermonkey.conf.xml.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Zhenya Malyshev
Skickat: den 25 september 2012 16:26
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] [ERR] mod_spidermonkey.c:3439 Error loading ODBC, followed by ReferenceError: ODBC is not defined

Hello everyone,

I am trying to resolve the issue that was discussed a few times in this user group, but by some reason recommended solutions do not work for me.

While calling MySQL using ODBC from javascript I am getting the famous [ERR] mod_spidermonkey.c:3439 Error loading ODBC, followed by ReferenceError: ODBC is not defined
I searched for it and found a perfect explanation by Michael J. and other group participants how to resolve it:

http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006863.html
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc

I installed unixODBC and found <load mod_spidermonkey_odbc /> in spidermonkey.conf.xml,
but I cannot uncomment the languages/mod_spidermonkey_odbc line within the modules.conf file since there is no such line.
I tried to add it, but got the error while re-compiling. While browsing through this user group discussions I sometimes notice inaccuracy in
recommendations which are no essential for experienced users, but misleading for armatures like myself.

I would greatly appreciate if someone could clarify that languages/mod_spidermonkey_odbc in modules.conf statement.
There is only one line referencing spidermonkey and it is languages/mod_spidermonkey.

Thanks,

Eugene
!DSPAM:5061f2c932768342020278!
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From peter.olsson at visionutveckling.se  Wed Sep 26 12:26:59 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 26 Sep 2012 08:26:59 +0000
Subject: [Freeswitch-users] Sometimes missing a few seconds of audio when CN
 is offered in INVITE (and solution)
Message-ID: <1FFF97C269757C458224B7C895F35F1515D7FA@cantor.std.visionutv.se>

Hello everyone! I experience a strange issue for bridged calls from Lync -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I'm not sure if this a bug or not so I ask the question here first.

The problem is that the audio back from the Asterisk server is not bridged back to the originating client for the first few seconds (how long is different on different calls), instead FS seems to send CN-packets back. I've looked into a wireshark dump, and I see the audio is coming from the Asterisk server (using the correct IP and port), but FS doesn't write the same packet to the other call leg (back to Lync), instead it sends a CN packet every second or so. My guess is that the timestamp is being handled wrong somehow, and FS believes that the real packet is too old (since it has already sent a CN packet with a higher timestamp?), and should not be written to the other leg, then after a while it sends a new CN packet and so on.

The solution is to set  "suppress_cng=true" before bridging the call to the Asterisk server, when this is done, the audio is always bridged correctly, and nothing is missing.

I have pcaps of both working calls (with the variable set) and nonworking calls, so if you believe it might be something that FS should handle differently I'll submit this to Jira.

/Peter
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From x.liu at hw.ac.uk  Wed Sep 26 14:33:35 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Wed, 26 Sep 2012 11:33:35 +0100
Subject: [Freeswitch-users] Is the FS JIRA server not working properly
 now?
In-Reply-To: <CC7616F1.4D038%krice@freeswitch.org>
References: <CC7616F1.4D038%krice@freeswitch.org>
Message-ID: <5062D9FF.9060109@hw.ac.uk>

Hi,

I am wondering if the Jira server is running normally or not now.

I still do not get notification via an email for my Jira comments added 
yesterday.

The issue I reported was hanging (no progress update received) there for 
a while, which makes me wonder
even if the Assignee could get the message from the Jira.
(My Jira report: FS-4527: mod_rtmp audio codecs issues for speech 
recognition?)

Thanks!
Xing


On 09/12/2012 04:39 PM, Ken Rice wrote:
> Its possible...we just did an upgrade of Jira... I'll check it out...
>
> K
>
>
> On 9/12/12 9:20 AM, "x.liu"<x.liu at hw.ac.uk>  wrote:
>
>> Hi,
>>
>> Normally after I add a comment to an existing issue I will receive an
>> email copy of the comment very soon.
>>
>> I haven't received the email after I added a comment 6 hours ago today.
>> so I am wondering if there is something wrong.
>>
>> 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.



From x.liu at hw.ac.uk  Wed Sep 26 14:38:32 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Wed, 26 Sep 2012 11:38:32 +0100
Subject: [Freeswitch-users] Is the FS JIRA server not working properly
 now?
In-Reply-To: <CC7616F1.4D038%krice@freeswitch.org>
References: <CC7616F1.4D038%krice@freeswitch.org>
Message-ID: <5062DB28.4040200@hw.ac.uk>

Hi,

I am wondering if the Jira server is running normally or not now.

I still do not get notification via an email for my Jira comments added 
yesterday.

The issue I reported was hanging (no progress update received) there for 
a while, which makes me wonder
even if the Assignee could get the message from the Jira.
(My Jira report: FS-4527: mod_rtmp audio codecs issues for speech 
recognition?)

Thanks!
Xing





On 09/12/2012 04:39 PM, Ken Rice wrote:
> Its possible...we just did an upgrade of Jira... I'll check it out...
>
> K
>
>
> On 9/12/12 9:20 AM, "x.liu"<x.liu at hw.ac.uk>  wrote:
>
>> Hi,
>>
>> Normally after I add a comment to an existing issue I will receive an
>> email copy of the comment very soon.
>>
>> I haven't received the email after I added a comment 6 hours ago today.
>> so I am wondering if there is something wrong.
>>
>> 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.



From david.villasmil.work at gmail.com  Wed Sep 26 17:33:29 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 26 Sep 2012 15:33:29 +0200
Subject: [Freeswitch-users] CLI Debug request?
In-Reply-To: <CAD-qRkiz6AA6G3qqGWHx=PodFfN4OyqJHu7pBHPdWtJ4mgZazQ@mail.gmail.com>
References: <CAFGRPVr6KgMibcit3r2-s0v--HKiO5s6NtomG9V6P6qCj=o4vA@mail.gmail.com>
	<CAD-qRkiz6AA6G3qqGWHx=PodFfN4OyqJHu7pBHPdWtJ4mgZazQ@mail.gmail.com>
Message-ID: <CAFGRPVopvDky20EHygWipmE702YkR5_QSS_-tQrYAxzJaC7bsg@mail.gmail.com>

Thanks for the answers!

Christopher, that's EXACTLY what I needed, Thanks!


David

On Wed, Sep 26, 2012 at 4:11 AM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> There is a session_loglevel application you can call from the dialplan.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_session_loglevel
>
>
>
> On Tue, Sep 25, 2012 at 5:52 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello All,
>>
>> It would be VERY nice to be able to set a "fsctl loglevel debug" on a
>> specific "called number", maybe by looking at the cldnum and filtering all
>> output but the uniqueID?
>> And by this I mean Sofia as much as ANY log information happening ONLY
>> for that specific call..
>>
>> When one has high loads it is extremely difficult to debug a specific
>> situation...
>>
>> Maybe this is already possible, but I don't know how?
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Sep 26 18:04:23 2012
From: tahir at ictinnovations.com (Tahir Almas)
Date: Wed, 26 Sep 2012 19:04:23 +0500
Subject: [Freeswitch-users] Video Calls over Freeswitch
Message-ID: <CAOL0t++c+6vqPMa-=-puuB-cx25kt2RzFdZu7eoWPfDQCeeVXQ@mail.gmail.com>

I will appreciate any help regarding video call support in Freeswitch and
how it can be implemented using  Andriod phones ?


*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
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From gerald.weber at besharp.at  Wed Sep 26 19:06:40 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Wed, 26 Sep 2012 15:06:40 +0000
Subject: [Freeswitch-users] intercom=true on originate bleg
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2A583F@exchange.besharp.at>

Hi,

is there a reason why intercom=true is sent to the bleg (2001) with

originate [sip_auto_answer=true,execute_on_media='set sip_auto_answer=']user/2000 2001
(from http://jira.freeswitch.org/browse/FS-3223)
or
originate [sip_auto_answer=true]user/2000 &bridge([sip_auto_answer=false]user/2001)
(from http://lists.freeswitch.org/pipermail/freeswitch-users/2012-January/079524.html)

?

2000 is a snom phone which auto answers correctly, 2001 is phonerlite.

I thought its possible to turn that off with sip_auto_answer=false
Wireshark trace is here http://pastebin.freeswitch.com/19938
Sofia loglevel all 9 and tracelevel debug is here http://pastebin.freeswitch.com/19939
I'm on latest git
FreeSWITCH Version 1.3.0+git~20120925T211258Z~817439d76a (git 817439d 2012-09-25 21:12:58Z)


Thx
gw
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From msc at freeswitch.org  Wed Sep 26 20:11:38 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Sep 2012 09:11:38 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call Today
Message-ID: <CAKzWOxW__ATtuWKL5bkMrP_r9d-2PQACXmXN80bSzMt4q+t23Q@mail.gmail.com>

Don't forget about today's call!

http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26

We get to have a deeper technical conversation with Scott and Daniel
regarding the repro SIP proxy. 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 shahzad.bhatti at g-r-v.com  Wed Sep 26 20:36:27 2012
From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti)
Date: Wed, 26 Sep 2012 21:36:27 +0500
Subject: [Freeswitch-users] get events data on freeswitch console
Message-ID: <CAN+NqUZnVvm18x7K7rZ+7mt6qphmVuwUetXnPhdzT_tGU84oeA@mail.gmail.com>

hi,

i want to get the events data without ESL in lua and with the following:

freeswitch.consoleLog("info",params:serialize())


but unable to any output but* ERROR*

2012-09-27 01:21:47.024894 [ERR] mod_lua.cpp:198
/usr/local/freeswitch/scripts/ESLInLua.lua:5: attempt to index global
'params' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/ESLInLua.lua:5: in main chunk


reply me how i can get that.

Regards

Shahzad Bhatti
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From miconda at gmail.com  Wed Sep 26 21:12:08 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 26 Sep 2012 19:12:08 +0200
Subject: [Freeswitch-users] voicebox email notifications on Debian with
 exim4
In-Reply-To: <CAKV-jFzQdV5biKpWagccJ0CjO5hXUaSMAD77sa_LZC1KRH948A@mail.gmail.com>
References: <50604618.4010602@gmail.com>
	<CAKV-jFzQdV5biKpWagccJ0CjO5hXUaSMAD77sa_LZC1KRH948A@mail.gmail.com>
Message-ID: <50633768.3060601@gmail.com>

I already added to the wiki by the time you replied (also sent an update 
to the mailing list). Before that, msmtp was mentioned only for windows.

You can find what I added at:
   * 
http://wiki.freeswitch.org/wiki/Mod_voicemail#Using_MSMTP_for_Local_Relay_to_Exim4_on_Debian 


Plus a note at beginning of:
   * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings

Cheers,
Daniel

On 9/24/12 11:12 PM, Brian Foster wrote:
>
> Im pretty sure this is mentioned already on the wiki, but please 
> elaborate your process to fill in some gaps.
>
> -BDF
>
> On Sep 24, 2012 8:33 AM, "Daniel-Constantin Mierla" <miconda at gmail.com 
> <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     I got into the crashing issue for freeswitch voicemail email
>     notifications on Debian with exim4, reported at:
>        * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>
>     My solution was to use msmtp alongside exim4 -- msmtp is used to relay
>     the emails to the local exim4 instance. All seems to work ok, I wonder
>     if anyone else is using same approach and got any issues or there are
>     other solutions found meanwhile for debian+exim4 combination.
>
>     I plan to put more details there, once the registration process for my
>     account on the wiki is completed.
>
>     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
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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From anthony.minessale at gmail.com  Wed Sep 26 21:45:51 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Sep 2012 12:45:51 -0500
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <5062A92A.2090808@gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
	<5062A92A.2090808@gmail.com>
Message-ID: <CAKbxfG9b1e24BoCW6fi54uVaXtzDQcWDZa2y5SjzPBKr4se1sg@mail.gmail.com>

What method are you using to record?

On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj at gmail.com> wrote:

> On 26.09.2012 04:55, Tiago Sampaio wrote:
> > I tested with 1.3.0, and the problem still occours...
> >
> > Testing to better debug the issue, i can reach this:
> >
> > PSTN -> FS -> user/1000 (record ok!)
> > user/1000 -> FS -> user/1001 (record ok!)
> > user/1000 -> FS -> PSTN problem occurs..
>
> So problem occur when an VoIP client calls the PSTN client, ie an
> outbound call from FS to PSTN. Checkout in vars.xml the option
> outbound_codec_prefs and it must contain the same codecs, especialy PCMU
> or PCMA.
>
> --
> 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
>



-- 
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 tnsampaio at bsd.com.br  Wed Sep 26 22:01:14 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Wed, 26 Sep 2012 15:01:14 -0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <CAKbxfG9b1e24BoCW6fi54uVaXtzDQcWDZa2y5SjzPBKr4se1sg@mail.gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
	<5062A92A.2090808@gmail.com>
	<CAKbxfG9b1e24BoCW6fi54uVaXtzDQcWDZa2y5SjzPBKr4se1sg@mail.gmail.com>
Message-ID: <506342EA.5020106@bsd.com.br>

This is an example...

                 <extension name="numero_local">
                         <condition field="destination_number" 
expression="^(\d{8})$">
                                 <action application="bind_meta_app" 
data="3 a a execute_extension::att_xfer XML features"/>
                                 <action application="set" 
data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} 
${strftime(%d/%m/%Y %H:%M)}"/>
                                 <action application="set" 
data="RECORD_COPYRIGHT=(c) SAMGAR"/>
                                 <action application="set" 
data="RECORD_SOFTWARE=SGIPBX v1.0"/>
                                 <action application="set" 
data="RECORD_ARTIST=SAMGAR"/>
                                 <action application="set" 
data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/>
                                 <action application="set" 
data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/>
                                 <action application="set" 
data="RECORD_STEREO=true"/>
                                 <action application="set" 
data="media_bug_answer_req=true"/>
                                 <action application="record_session" 
data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/>
                                 <action application="bridge" 
data="Khomp/b0/$1" />
                         </condition>
                 </extension>

in any format issue occurs...

Em 26/09/2012 14:45, Anthony Minessale escreveu:
> What method are you using to record?
>
> On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj at gmail.com 
> <mailto:vbvbrj at gmail.com>> wrote:
>
>     On 26.09.2012 04 <tel:26.09.2012%2004>:55, Tiago Sampaio wrote:
>     > I tested with 1.3.0, and the problem still occours...
>     >
>     > Testing to better debug the issue, i can reach this:
>     >
>     > PSTN -> FS -> user/1000 (record ok!)
>     > user/1000 -> FS -> user/1001 (record ok!)
>     > user/1000 -> FS -> PSTN problem occurs..
>
>     So problem occur when an VoIP client calls the PSTN client, ie an
>     outbound call from FS to PSTN. Checkout in vars.xml the option
>     outbound_codec_prefs and it must contain the same codecs,
>     especialy PCMU
>     or PCMA.
>
>     --
>     Mimiko desu.
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 26 22:28:04 2012
From: toddb at toddbailey.net (Todd Bailey)
Date: Wed, 26 Sep 2012 11:28:04 -0700
Subject: [Freeswitch-users] Hacking FS issue
Message-ID: <1348684084.7087.16.camel@mythtv.toddbailey.net>


Hey All,


I just got an email from Frontier that there were several attempts to
make international calls.


I checked the log file and verified that somehow someone was able to get
access to FS from the internet. 


here is a sample of the log

2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
Channel sofia/internal/1000 at 50.47.85.167
[af778857-0188-4ed2-a82a-94ae749a02cb]
2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
Processing 1000 <1000>->01137168521352 in context default
2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
Channel sofia/internal/01137168521352 at 192.168.1.5:5061
[d1243a78-c464-45fa-9215-e7b85e1221fc]
2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
sofia/internal/01137168521352 at 192.168.1.5:5061!
2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
sofia/internal/1000 at 50.47.85.167!
2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
Ring Ready sofia/internal/1000 at 50.47.85.167!
2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
[sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/internal/1000 at 50.47.85.167!
2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
Channel [sofia/internal/1000 at 50.47.85.167] has been answered
2012-09-23 16:30:52.356865 [N2012-09-23 16:30:29.916821
[NOTICE] switch_channel.c:941 New Channel
sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
Processing 1000 <1000>->01137168521352 in context default
2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
Channel sofia/internal/01137168521352 at 192.168.1.5:5061
[d1243a78-c464-45fa-9215-e7b85e1221fc]
2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
sofia/internal/01137168521352 at 192.168.1.5:5061!
2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
sofia/internal/1000 at 50.47.85.167!
2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
Ring Ready sofia/internal/1000 at 50.47.85.167!
2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
[sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/internal/1000 at 50.47.85.167!
2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
Channel [sofia/internal/1000 at 50.47.85.167] has been answered
2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
Channel sofia/internal/1000 at 50.47.85.167
[4576bc76-144a-4f6f-8915-871b511c374d]
2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
Processing 1000 <1000>->01137168905352 in context defaultOTICE]
switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
[4576bc76-144a-4f6f-8915-871b511c374d]
2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
Processing 1000 <1000>->01137168905352 in context default


At this point I'm at a loss how this is happening as I have multiple
firewalls in place that limit port access.

Can someone provide a few pointers on how to better secure FS running on
Linux systems?


thanks


-- 
-
-
-    Best Regards,
-
-            Todd Bailey
-
-



From anthony.minessale at gmail.com  Wed Sep 26 22:30:13 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Sep 2012 13:30:13 -0500
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <506342EA.5020106@bsd.com.br>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
	<5062A92A.2090808@gmail.com>
	<CAKbxfG9b1e24BoCW6fi54uVaXtzDQcWDZa2y5SjzPBKr4se1sg@mail.gmail.com>
	<506342EA.5020106@bsd.com.br>
Message-ID: <CAKbxfG8ms8H2g8uZrHaUGnbY-Fxd_M0QD4_ws40J=mHBWuy76A@mail.gmail.com>

Must be a khomp specific issue.  Maybe the frames coming from it are
reporting the wrong size or something...


On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio <tnsampaio at bsd.com.br>wrote:

>  This is an example...
>
>                 <extension name="numero_local">
>                         <condition field="destination_number"
> expression="^(\d{8})$">
>                                 <action application="bind_meta_app"
> data="3 a a execute_extension::att_xfer XML features"/>
>                                 <action application="set"
> data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number}
> ${strftime(%d/%m/%Y %H:%M)}"/>
>                                 <action application="set"
> data="RECORD_COPYRIGHT=(c) SAMGAR"/>
>                                 <action application="set"
> data="RECORD_SOFTWARE=SGIPBX v1.0"/>
>                                 <action application="set"
> data="RECORD_ARTIST=SAMGAR"/>
>                                 <action application="set"
> data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/>
>                                 <action application="set"
> data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/>
>                                 <action application="set"
> data="RECORD_STEREO=true"/>
>                                 <action application="set"
> data="media_bug_answer_req=true"/>
>                                 <action application="record_session"
> data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/>
>                                 <action application="bridge"
> data="Khomp/b0/$1" />
>                         </condition>
>                 </extension>
>
> in any format issue occurs...
>
> Em 26/09/2012 14:45, Anthony Minessale escreveu:
>
> What method are you using to record?
>
> On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
>
>> On 26.09.2012 04 <26.09.2012%2004>:55, Tiago Sampaio wrote:
>> > I tested with 1.3.0, and the problem still occours...
>> >
>> > Testing to better debug the issue, i can reach this:
>> >
>> > PSTN -> FS -> user/1000 (record ok!)
>> > user/1000 -> FS -> user/1001 (record ok!)
>> > user/1000 -> FS -> PSTN problem occurs..
>>
>>  So problem occur when an VoIP client calls the PSTN client, ie an
>> outbound call from FS to PSTN. Checkout in vars.xml the option
>> outbound_codec_prefs and it must contain the same codecs, especialy PCMU
>> or PCMA.
>>
>> --
>> 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
>>
>
>
>
>  --
> 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.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
>
>


-- 
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 avi at avimarcus.net  Wed Sep 26 22:37:05 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 26 Sep 2012 20:37:05 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <1348684084.7087.16.camel@mythtv.toddbailey.net>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
Message-ID: <CALNZuwmZE+c9Ur4dwYb1BcXj+Ee1uEU_W9Gat95jVUF9VE6DvQ@mail.gmail.com>

It looks like 50.47.85.167 auth'ed as user 1000. Seemingly, the internal
profile is exposed to the internet.

1) If you are purely running internal calls, then explicitly set the local
IP in the sofia profile so that it doesn't listen externally.
2) Did they auth as user 1000? Do you have that user set up?
3) Firewalls are best set up opposite, as blocking ALL inbound traffic then
explicitly punching hole to allow some. Are you allowing anything to do
with port 5060, 5061, etc or however you set up your sofia internal profile?

We need more info to tell you anything more...

-Avi


On Wed, Sep 26, 2012 at 8:28 PM, Todd Bailey <toddb at toddbailey.net> wrote:

>
> Hey All,
>
>
> I just got an email from Frontier that there were several attempts to
> make international calls.
>
>
> I checked the log file and verified that somehow someone was able to get
> access to FS from the internet.
>
>
> here is a sample of the log
>
>  [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [af778857-0188-4ed2-a82a-94ae749a02cb]
>  [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
>  [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>  [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
> [NOTICE] switch_channel.c:941 New Channel
> sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
>  [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
>  [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>  [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
>  [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
>  [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context default
>
>
> At this point I'm at a loss how this is happening as I have multiple
> firewalls in place that limit port access.
>
> Can someone provide a few pointers on how to better secure FS running on
> Linux systems?
>
>
> thanks
>
>
> --
> -
> -
> -    Best Regards,
> -
> -            Todd Bailey
> -
> -
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tnsampaio at bsd.com.br  Wed Sep 26 22:49:40 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Wed, 26 Sep 2012 15:49:40 -0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <CAKbxfG8ms8H2g8uZrHaUGnbY-Fxd_M0QD4_ws40J=mHBWuy76A@mail.gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
	<5062A92A.2090808@gmail.com>
	<CAKbxfG9b1e24BoCW6fi54uVaXtzDQcWDZa2y5SjzPBKr4se1sg@mail.gmail.com>
	<506342EA.5020106@bsd.com.br>
	<CAKbxfG8ms8H2g8uZrHaUGnbY-Fxd_M0QD4_ws40J=mHBWuy76A@mail.gmail.com>
Message-ID: <50634E44.5030602@bsd.com.br>

The problem is to debug it...
I didnt saw anything in logs or debug.....
Can you tell me a way to debug it?

Em 26/09/2012 15:30, Anthony Minessale escreveu:
> Must be a khomp specific issue.  Maybe the frames coming from it are 
> reporting the wrong size or something...
>
>
> On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio 
> <tnsampaio at bsd.com.br <mailto:tnsampaio at bsd.com.br>> wrote:
>
>     This is an example...
>
>                     <extension name="numero_local">
>                             <condition field="destination_number"
>     expression="^(\d{8})$">
>                                     <action
>     application="bind_meta_app" data="3 a a
>     execute_extension::att_xfer XML features"/>
>                                     <action application="set"
>     data="RECORD_TITLE=Recording ${destination_number}
>     ${caller_id_number} ${strftime(%d/%m/%Y %H:%M)}"/>
>                                     <action application="set"
>     data="RECORD_COPYRIGHT=(c) SAMGAR"/>
>                                     <action application="set"
>     data="RECORD_SOFTWARE=SGIPBX v1.0"/>
>                                     <action application="set"
>     data="RECORD_ARTIST=SAMGAR"/>
>                                     <action application="set"
>     data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/>
>                                     <action application="set"
>     data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/>
>                                     <action application="set"
>     data="RECORD_STEREO=true"/>
>                                     <action application="set"
>     data="media_bug_answer_req=true"/>
>                                     <action
>     application="record_session"
>     data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/>
>                                     <action application="bridge"
>     data="Khomp/b0/$1" />
>                             </condition>
>                     </extension>
>
>     in any format issue occurs...
>
>     Em 26/09/2012 14:45, Anthony Minessale escreveu:
>>     What method are you using to record?
>>
>>     On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj at gmail.com
>>     <mailto:vbvbrj at gmail.com>> wrote:
>>
>>         On 26.09.2012 04 <tel:26.09.2012%2004>:55, Tiago Sampaio wrote:
>>         > I tested with 1.3.0, and the problem still occours...
>>         >
>>         > Testing to better debug the issue, i can reach this:
>>         >
>>         > PSTN -> FS -> user/1000 (record ok!)
>>         > user/1000 -> FS -> user/1001 (record ok!)
>>         > user/1000 -> FS -> PSTN problem occurs..
>>
>>         So problem occur when an VoIP client calls the PSTN client, ie an
>>         outbound call from FS to PSTN. Checkout in vars.xml the option
>>         outbound_codec_prefs and it must contain the same codecs,
>>         especialy PCMU
>>         or PCMA.
>>
>>         --
>>         Mimiko desu.
>>
>>         _________________________________________________________________________
>>         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
>>
>>
>>
>>
>>     -- 
>>     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  <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
>
>
>
>
> -- 
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 26 22:53:21 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 26 Sep 2012 13:53:21 -0500
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <1348684084.7087.16.camel@mythtv.toddbailey.net>
Message-ID: <CC88B951.543A0%krice@freeswitch.org>


NormT at VoiceNetwork has some good stuff for this on the VoiceNetwork.ca
wiki (check him out for Orig and Term also!)

Fail2Ban can stops brute force attempts
http://wiki.voicenetwork.ca/wiki/Main_Page#Fail2Ban


IPTables Rules to help mitigate some brute force and DDoS attacks
http://wiki.voicenetwork.ca/wiki/Iptables


On 9/26/12 1:28 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:

> 
> Hey All,
> 
> 
> I just got an email from Frontier that there were several attempts to
> make international calls.
> 
> 
> I checked the log file and verified that somehow someone was able to get
> access to FS from the internet.
> 
> 
> here is a sample of the log
> 
> 2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [af778857-0188-4ed2-a82a-94ae749a02cb]
> 2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
> 2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
> 2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
> 2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
> 2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
> 2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
> 2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
> 2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
> 2012-09-23 16:30:52.356865 [N2012-09-23 16:30:29.916821
> [NOTICE] switch_channel.c:941 New Channel
> sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
> 2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
> 2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
> 2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
> 2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
> 2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
> 2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
> 2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
> 2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
> 2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
> 2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
> 2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context default
> 
> 
> At this point I'm at a loss how this is happening as I have multiple
> firewalls in place that limit port access.
> 
> Can someone provide a few pointers on how to better secure FS running on
> Linux systems?
> 
> 
> thanks
> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From shaheryarkh at googlemail.com  Wed Sep 26 22:59:36 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 26 Sep 2012 20:59:36 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <1348684084.7087.16.camel@mythtv.toddbailey.net>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
Message-ID: <CAFZQphw7_kBthuKddZhkStgXK0L56tD0h+qWrqwWOXAJ3UoxuQ@mail.gmail.com>

Fs comed with many sample accounts. There are at least 20 sample directory
users 1000-1020 which you should remove before putting it to production.

On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>
>
> Hey All,
>
>
> I just got an email from Frontier that there were several attempts to
> make international calls.
>
>
> I checked the log file and verified that somehow someone was able to get
> access to FS from the internet.
>
>
> here is a sample of the log
>
>  [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [af778857-0188-4ed2-a82a-94ae749a02cb]
>  [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
>  [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>  [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
> [NOTICE] switch_channel.c:941 New Channel
> sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
>  [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
>  [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>  [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
>  [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
>  [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context default
>
>
> At this point I'm at a loss how this is happening as I have multiple
> firewalls in place that limit port access.
>
> Can someone provide a few pointers on how to better secure FS running on
> Linux systems?
>
>
> thanks
>
>
> --
> -
> -
> -    Best Regards,
> -
> -            Todd Bailey
> -
> -
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bigx333 at gmail.com  Wed Sep 26 22:54:46 2012
From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado)
Date: Wed, 26 Sep 2012 20:54:46 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <1348684084.7087.16.camel@mythtv.toddbailey.net>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
Message-ID: <CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>

I really think that people give way too much importance to firewalls,
specially stateless ones, blocking ports isn't going to do much for you
unless you are trying to hide vulnerable services behind it.

They used the extension 1000 to make the calls so I would say: activate
log-auth-failures on your profile, setup a fail2ban and get stronger
passwords.

If you want to go further you can use a stateful firewall limiting
connections and setup a IDS(recommend snort)
On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:

>
> Hey All,
>
>
> I just got an email from Frontier that there were several attempts to
> make international calls.
>
>
> I checked the log file and verified that somehow someone was able to get
> access to FS from the internet.
>
>
> here is a sample of the log
>
>  [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [af778857-0188-4ed2-a82a-94ae749a02cb]
>  [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
>  [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>  [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
> [NOTICE] switch_channel.c:941 New Channel
> sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
>  [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168521352 in context default
>  [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> sofia/internal/01137168521352 at 192.168.1.5:5061!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> Ring Ready sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> sofia/internal/1000 at 50.47.85.167!
>  [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>  [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
> Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
>  [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
> [4576bc76-144a-4f6f-8915-871b511c374d]
>  [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> Processing 1000 <1000>->01137168905352 in context default
>
>
> At this point I'm at a loss how this is happening as I have multiple
> firewalls in place that limit port access.
>
> Can someone provide a few pointers on how to better secure FS running on
> Linux systems?
>
>
> thanks
>
>
> --
> -
> -
> -    Best Regards,
> -
> -            Todd Bailey
> -
> -
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed Sep 26 23:28:00 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Sep 2012 14:28:00 -0500
Subject: [Freeswitch-users] Sometimes missing a few seconds of audio
 when CN is offered in INVITE (and solution)
In-Reply-To: <1FFF97C269757C458224B7C895F35F1515D7FA@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1515D7FA@cantor.std.visionutv.se>
Message-ID: <CAKbxfG9MC_8GpJZo3giZx2EpvJtOyssTuqx2jqtGvRkgkM=tyw@mail.gmail.com>

Looking at the code, I can see a solution assuming your theory is correct.
I pushed a patch to HEAD if you care to test it.


On Wed, Sep 26, 2012 at 3:26 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Hello everyone! I experience a strange issue for bridged calls from Lync
> -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I?m not
> sure if this a bug or not so I ask the question here first.****
>
> ** **
>
> The problem is that the audio back from the Asterisk server is not bridged
> back to the originating client for the first few seconds (how long is
> different on different calls), instead FS seems to send CN-packets back.
> I?ve looked into a wireshark dump, and I see the audio is coming from the
> Asterisk server (using the correct IP and port), but FS doesn?t write the
> same packet to the other call leg (back to Lync), instead it sends a CN
> packet every second or so. My guess is that the timestamp is being handled
> wrong somehow, and FS believes that the real packet is too old (since it
> has already sent a CN packet with a higher timestamp?), and should not be
> written to the other leg, then after a while it sends a new CN packet and
> so on.****
>
> ** **
>
> The solution is to set  ?suppress_cng=true? before bridging the call to
> the Asterisk server, when this is done, the audio is always bridged
> correctly, and nothing is missing.****
>
> ** **
>
> I have pcaps of both working calls (with the variable set) and nonworking
> calls, so if you believe it might be something that FS should handle
> differently I?ll submit this to Jira.****
>
> ** **
>
> /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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Wed Sep 26 23:52:12 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Sep 2012 14:52:12 -0500
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <50634E44.5030602@bsd.com.br>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
	<5062A92A.2090808@gmail.com>
	<CAKbxfG9b1e24BoCW6fi54uVaXtzDQcWDZa2y5SjzPBKr4se1sg@mail.gmail.com>
	<506342EA.5020106@bsd.com.br>
	<CAKbxfG8ms8H2g8uZrHaUGnbY-Fxd_M0QD4_ws40J=mHBWuy76A@mail.gmail.com>
	<50634E44.5030602@bsd.com.br>
Message-ID: <CAKbxfG9LUU3vbmeQE3R0mu61XAL456ir=JHbMPH3ddDac_ZJfQ@mail.gmail.com>

maybe edit the code and print out values like the frame datalen or samples
in the read_frame function in the mod and see if you can see anything odd.


On Wed, Sep 26, 2012 at 1:49 PM, Tiago N. Sampaio <tnsampaio at bsd.com.br>wrote:

>  The problem is to debug it...
> I didnt saw anything in logs or debug.....
> Can you tell me a way to debug it?
>
> Em 26/09/2012 15:30, Anthony Minessale escreveu:
>
> Must be a khomp specific issue.  Maybe the frames coming from it are
> reporting the wrong size or something...
>
>
> On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio <tnsampaio at bsd.com.br>wrote:
>
>>  This is an example...
>>
>>                 <extension name="numero_local">
>>                         <condition field="destination_number"
>> expression="^(\d{8})$">
>>                                 <action application="bind_meta_app"
>> data="3 a a execute_extension::att_xfer XML features"/>
>>                                 <action application="set"
>> data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number}
>> ${strftime(%d/%m/%Y %H:%M)}"/>
>>                                 <action application="set"
>> data="RECORD_COPYRIGHT=(c) SAMGAR"/>
>>                                 <action application="set"
>> data="RECORD_SOFTWARE=SGIPBX v1.0"/>
>>                                 <action application="set"
>> data="RECORD_ARTIST=SAMGAR"/>
>>                                 <action application="set"
>> data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/>
>>                                 <action application="set"
>> data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/>
>>                                 <action application="set"
>> data="RECORD_STEREO=true"/>
>>                                 <action application="set"
>> data="media_bug_answer_req=true"/>
>>                                 <action application="record_session"
>> data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/>
>>                                 <action application="bridge"
>> data="Khomp/b0/$1" />
>>                         </condition>
>>                 </extension>
>>
>> in any format issue occurs...
>>
>> Em 26/09/2012 14:45, Anthony Minessale escreveu:
>>
>> What method are you using to record?
>>
>> On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
>>
>>> On 26.09.2012 04 <26.09.2012%2004>:55, Tiago Sampaio wrote:
>>> > I tested with 1.3.0, and the problem still occours...
>>> >
>>> > Testing to better debug the issue, i can reach this:
>>> >
>>> > PSTN -> FS -> user/1000 (record ok!)
>>> > user/1000 -> FS -> user/1001 (record ok!)
>>> > user/1000 -> FS -> PSTN problem occurs..
>>>
>>>  So problem occur when an VoIP client calls the PSTN client, ie an
>>> outbound call from FS to PSTN. Checkout in vars.xml the option
>>> outbound_codec_prefs and it must contain the same codecs, especialy PCMU
>>> or PCMA.
>>>
>>> --
>>> 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
>>>
>>
>>
>>
>>  --
>> 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.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
>>
>>
>
>
>  --
> 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.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
>
>


-- 
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 peter.olsson at visionutveckling.se  Wed Sep 26 23:56:07 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 26 Sep 2012 19:56:07 +0000
Subject: [Freeswitch-users] Sometimes missing a few seconds of audio
 when CN is offered in INVITE (and solution)
In-Reply-To: <CAKbxfG9MC_8GpJZo3giZx2EpvJtOyssTuqx2jqtGvRkgkM=tyw@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1515D7FA@cantor.std.visionutv.se>,
	<CAKbxfG9MC_8GpJZo3giZx2EpvJtOyssTuqx2jqtGvRkgkM=tyw@mail.gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15160853@cantor.std.visionutv.se>

Thanks Tony, I will try it out tomorrow - I'll get back with the results!

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 26 september 2012 21:28
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution)

Looking at the code, I can see a solution assuming your theory is correct.
I pushed a patch to HEAD if you care to test it.


On Wed, Sep 26, 2012 at 3:26 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Hello everyone! I experience a strange issue for bridged calls from Lync -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I?m not sure if this a bug or not so I ask the question here first.

The problem is that the audio back from the Asterisk server is not bridged back to the originating client for the first few seconds (how long is different on different calls), instead FS seems to send CN-packets back. I?ve looked into a wireshark dump, and I see the audio is coming from the Asterisk server (using the correct IP and port), but FS doesn?t write the same packet to the other call leg (back to Lync), instead it sends a CN packet every second or so. My guess is that the timestamp is being handled wrong somehow, and FS believes that the real packet is too old (since it has already sent a CN packet with a higher timestamp?), and should not be written to the other leg, then after a while it sends a new CN packet and so on.

The solution is to set  ?suppress_cng=true? before bridging the call to the Asterisk server, when this is done, the audio is always bridged correctly, and nothing is missing.

I have pcaps of both working calls (with the variable set) and nonworking calls, so if you believe it might be something that FS should handle differently I?ll submit this to Jira.

/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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at 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
!DSPAM:5063553532761904613329!


From tnsampaio at bsd.com.br  Thu Sep 27 00:34:26 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Wed, 26 Sep 2012 17:34:26 -0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <CAKbxfG9LUU3vbmeQE3R0mu61XAL456ir=JHbMPH3ddDac_ZJfQ@mail.gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<5061EA74.5040001@bsd.com.br> <5061F423.8020301@gmail.com>
	<CAJxjjuvrQSHGiLYZeg7gbEyc8u9qMSD9kZK+MrrSC6qNeBk+JA@mail.gmail.com>
	<5062A92A.2090808@gmail.com>
	<CAKbxfG9b1e24BoCW6fi54uVaXtzDQcWDZa2y5SjzPBKr4se1sg@mail.gmail.com>
	<506342EA.5020106@bsd.com.br>
	<CAKbxfG8ms8H2g8uZrHaUGnbY-Fxd_M0QD4_ws40J=mHBWuy76A@mail.gmail.com>
	<50634E44.5030602@bsd.com.br>
	<CAKbxfG9LUU3vbmeQE3R0mu61XAL456ir=JHbMPH3ddDac_ZJfQ@mail.gmail.com>
Message-ID: <506366D2.1020706@bsd.com.br>

Now if i use speex at 8000 it records normally now...
I will let it as is for now!
On next weekend i will try this to get more information!!

Thx for help!

Em 26/09/2012 16:52, Anthony Minessale escreveu:
> maybe edit the code and print out values like the frame datalen or 
> samples in the read_frame function in the mod and see if you can see 
> anything odd.
>
>
> On Wed, Sep 26, 2012 at 1:49 PM, Tiago N. Sampaio 
> <tnsampaio at bsd.com.br <mailto:tnsampaio at bsd.com.br>> wrote:
>
>     The problem is to debug it...
>     I didnt saw anything in logs or debug.....
>     Can you tell me a way to debug it?
>
>     Em 26/09/2012 15:30, Anthony Minessale escreveu:
>>     Must be a khomp specific issue.  Maybe the frames coming from it
>>     are reporting the wrong size or something...
>>
>>
>>     On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio
>>     <tnsampaio at bsd.com.br <mailto:tnsampaio at bsd.com.br>> wrote:
>>
>>         This is an example...
>>
>>                         <extension name="numero_local">
>>                                 <condition field="destination_number"
>>         expression="^(\d{8})$">
>>                                         <action
>>         application="bind_meta_app" data="3 a a
>>         execute_extension::att_xfer XML features"/>
>>                                         <action application="set"
>>         data="RECORD_TITLE=Recording ${destination_number}
>>         ${caller_id_number} ${strftime(%d/%m/%Y %H:%M)}"/>
>>                                         <action application="set"
>>         data="RECORD_COPYRIGHT=(c) SAMGAR"/>
>>                                         <action application="set"
>>         data="RECORD_SOFTWARE=SGIPBX v1.0"/>
>>                                         <action application="set"
>>         data="RECORD_ARTIST=SAMGAR"/>
>>                                         <action application="set"
>>         data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/>
>>                                         <action application="set"
>>         data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/>
>>                                         <action application="set"
>>         data="RECORD_STEREO=true"/>
>>                                         <action application="set"
>>         data="media_bug_answer_req=true"/>
>>                                         <action
>>         application="record_session"
>>         data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/>
>>                                         <action application="bridge"
>>         data="Khomp/b0/$1" />
>>                                 </condition>
>>                         </extension>
>>
>>         in any format issue occurs...
>>
>>         Em 26/09/2012 14:45, Anthony Minessale escreveu:
>>>         What method are you using to record?
>>>
>>>         On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj at gmail.com
>>>         <mailto:vbvbrj at gmail.com>> wrote:
>>>
>>>             On 26.09.2012 04 <tel:26.09.2012%2004>:55, Tiago Sampaio
>>>             wrote:
>>>             > I tested with 1.3.0, and the problem still occours...
>>>             >
>>>             > Testing to better debug the issue, i can reach this:
>>>             >
>>>             > PSTN -> FS -> user/1000 (record ok!)
>>>             > user/1000 -> FS -> user/1001 (record ok!)
>>>             > user/1000 -> FS -> PSTN problem occurs..
>>>
>>>             So problem occur when an VoIP client calls the PSTN
>>>             client, ie an
>>>             outbound call from FS to PSTN. Checkout in vars.xml the
>>>             option
>>>             outbound_codec_prefs and it must contain the same
>>>             codecs, especialy PCMU
>>>             or PCMA.
>>>
>>>             --
>>>             Mimiko desu.
>>>
>>>             _________________________________________________________________________
>>>             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
>>>
>>>
>>>
>>>
>>>         -- 
>>>         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  <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
>>
>>
>>
>>
>>     -- 
>>     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  <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
>
>
>
>
> -- 
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 27 01:19:05 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 26 Sep 2012 17:19:05 -0400
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
Message-ID: <0AC72351-CABE-41E7-A833-D987748373DB@kavun.ch>

Hi guys,

I am comparing this with an Asterisk and FreeSWITCH installation, using the same route, same codecs, same carrier, same phones and same servers? :P
I experience a delay when pressing DTMFs on the line that uses FreeSWITCH. I am estimating the delay to be around 500 ms.

What are the settings I can fine tune to avoid this?

All the best,
Emrah

From cesar.bermudez at gmail.com  Thu Sep 27 01:56:00 2012
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Wed, 26 Sep 2012 15:56:00 -0600
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <0AC72351-CABE-41E7-A833-D987748373DB@kavun.ch>
References: <0AC72351-CABE-41E7-A833-D987748373DB@kavun.ch>
Message-ID: <CADrcvV9XaUC+BG4weEKss1yTHWqGYesA0Go32EP4M66cLtRLAg@mail.gmail.com>

You tried this:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF

On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:

> Hi guys,
>
> I am comparing this with an Asterisk and FreeSWITCH installation, using
> the same route, same codecs, same carrier, same phones and same servers? :P
> I experience a delay when pressing DTMFs on the line that uses FreeSWITCH.
> I am estimating the delay to be around 500 ms.
>
> What are the settings I can fine tune to avoid this?
>
> All the 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 lists at kavun.ch  Thu Sep 27 02:44:28 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 26 Sep 2012 18:44:28 -0400
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <CADrcvV9XaUC+BG4weEKss1yTHWqGYesA0Go32EP4M66cLtRLAg@mail.gmail.com>
References: <0AC72351-CABE-41E7-A833-D987748373DB@kavun.ch>
	<CADrcvV9XaUC+BG4weEKss1yTHWqGYesA0Go32EP4M66cLtRLAg@mail.gmail.com>
Message-ID: <D9B39FD4-78E7-42DE-917D-3CAC3E25AF90@kavun.ch>

Yes I did.
BTW, the example in the Wiki contradicts the inline documentation in switch.xml.
    <!-- 
         The min-dtmf-duration specifies the minimum DTMF duration to use on 
         outgoing events. Events shorter than this will be increased in duration
         to match min_dtmf_duration. You cannot configure a dtmf duration on a 
         profile that is less than this setting. You may increase this value,
         but cannot set it lower than 400. This value cannot exceed 
         max-dtmf-duration. -->
The Wiki shows an example with the value at 100.

I tried increasing and decreasing it to no avail, it does not seem to interfere with anything I can measure with my ear. :P
On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com> wrote:

> You tried this: 
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
> 
> On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
> Hi guys,
> 
> I am comparing this with an Asterisk and FreeSWITCH installation, using the same route, same codecs, same carrier, same phones and same servers? :P
> I experience a delay when pressing DTMFs on the line that uses FreeSWITCH. I am estimating the delay to be around 500 ms.
> 
> What are the settings I can fine tune to avoid this?
> 
> All the 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lconroy at insensate.co.uk  Thu Sep 27 03:09:50 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Thu, 27 Sep 2012 00:09:50 +0100
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
Message-ID: <A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>

Hi There,
 welcome to our world; hope it didn't cost too much.
Frontier were pro-active, which is very good. Don't forget to thank them.
I'd guess that this particular bunch are coming from IP addresses provided in the West bank and/or Gaza; that's from where my "visitors" appeared to originate.

1st rule of fight club: Firewalls are no use for a server that is going to listen for requests from the Internet and allow authenticated calls to be placed from any IP address.

You MUST have reasonable passwords, plus fail2ban is easy to set up and works just fine [unless you're using Windoz, in which case God hates you**].

For more refined control (if you know where your external contacts are coming from) ...

Consider setting up ACLs (nailing down the IP address ranges from which you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition there is one place to add in your external correspondents.

Also, consider using cidr= parameters in your directory folder for each of your users (if they will only attempt to register or place calls from given address ranges).
Then enable ACLs for incalls in your sip profile(s).

This is all covered on wiki.freeswitch.org -- search for ACLs and take it from there.

BTW, you WILL be confused by setting explicit ACLs on registration -- leave that one commented out until you know what it actually does, as it's probably not what you expect. Several strong cups of coffee and protracted meditation may help.

Main message:
-- Immediately - fix the passwords so they're not easy to guess [as the bad guys *will* try again and again until they get it right].
-- set up fail2ban (which has its own page on the wiki) exactly as proposed. <======= MOST IMPORTANT
-- lose the belief that firewalls are going to help protect an Internet-listening server as, logically, they can't
Finally, be amazed at the occasional "block" reports in the fail2ban logfile, and wonder how you got away with it for so long.

all the best,
  Lawrence
** There was apparently a talk on how Windows users could get something close to a fail2ban-style setup (IIRC, it was on the weekly conf call a while back)

On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
> I really think that people give way too much importance to firewalls,
> specially stateless ones, blocking ports isn't going to do much for you
> unless you are trying to hide vulnerable services behind it.
> 
> They used the extension 1000 to make the calls so I would say: activate
> log-auth-failures on your profile, setup a fail2ban and get stronger
> passwords.
> 
> If you want to go further you can use a stateful firewall limiting
> connections and setup a IDS(recommend snort)
> On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
> 
>> 
>> Hey All,
>> 
>> 
>> I just got an email from Frontier that there were several attempts to
>> make international calls.
>> 
>> 
>> I checked the log file and verified that somehow someone was able to get
>> access to FS from the internet.
>> 
>> 
>> here is a sample of the log
>> 
>> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>> Channel sofia/internal/1000 at 50.47.85.167
>> [af778857-0188-4ed2-a82a-94ae749a02cb]
>> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>> Processing 1000 <1000>->01137168521352 in context default
>> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>> sofia/internal/01137168521352 at 192.168.1.5:5061!
>> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>> sofia/internal/1000 at 50.47.85.167!
>> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>> Ring Ready sofia/internal/1000 at 50.47.85.167!
>> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
>> sofia/internal/1000 at 50.47.85.167!
>> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
>> [NOTICE] switch_channel.c:941 New Channel
>> sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
>> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>> Processing 1000 <1000>->01137168521352 in context default
>> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>> sofia/internal/01137168521352 at 192.168.1.5:5061!
>> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>> sofia/internal/1000 at 50.47.85.167!
>> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>> Ring Ready sofia/internal/1000 at 50.47.85.167!
>> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
>> sofia/internal/1000 at 50.47.85.167!
>> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>> Channel sofia/internal/1000 at 50.47.85.167
>> [4576bc76-144a-4f6f-8915-871b511c374d]
>> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>> [4576bc76-144a-4f6f-8915-871b511c374d]
>> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>> Processing 1000 <1000>->01137168905352 in context default
>> 
>> 
>> At this point I'm at a loss how this is happening as I have multiple
>> firewalls in place that limit port access.
>> 
>> Can someone provide a few pointers on how to better secure FS running on
>> Linux systems?
>> 
>> 
>> thanks
>> 
>> 
>> --
>> -
>> -
>> -    Best Regards,
>> -
>> -            Todd Bailey
>> -
>> -
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Sep 27 06:34:25 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 26 Sep 2012 22:34:25 -0400
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <50614C83.1070300@gmail.com>
References: <5060B7C5.1060003@bsd.com.br>
	<50614C83.1070300@gmail.com>
Message-ID: <CAKV-jFw7AVzYKUY+sGume+ZKd=6A=a6_+P9QmemM6VQQsK91_Q@mail.gmail.com>

>
>
>
> Also checkout VoIP phone's settings. If you see "signaling standard" is
> set to "Chinese" change it to some European standard.
>
> Ah, that's why the Chinese talk so fast... :-)


-- 
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
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From vbvbrj at gmail.com  Thu Sep 27 09:36:55 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Thu, 27 Sep 2012 08:36:55 +0300
Subject: [Freeswitch-users] Recording problem speed and noise
In-Reply-To: <CAKV-jFw7AVzYKUY+sGume+ZKd=6A=a6_+P9QmemM6VQQsK91_Q@mail.gmail.com>
References: <5060B7C5.1060003@bsd.com.br> <50614C83.1070300@gmail.com>
	<CAKV-jFw7AVzYKUY+sGume+ZKd=6A=a6_+P9QmemM6VQQsK91_Q@mail.gmail.com>
Message-ID: <5063E5F7.7000605@gmail.com>

On 27.09.2012 05:34, Brian Foster wrote:
>
>
>     Also checkout VoIP phone's settings. If you see "signaling standard" is
>     set to "Chinese" change it to some European standard.
>
> Ah, that's why the Chinese talk so fast... :-)

Actually, it indeed resolved the problem for VoIP phones made in China.

-- 
Mimiko desu.


From peter.olsson at visionutveckling.se  Thu Sep 27 11:03:59 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 27 Sep 2012 07:03:59 +0000
Subject: [Freeswitch-users] Sometimes missing a few seconds of audio
 when CN is offered in INVITE (and solution)
Message-ID: <1FFF97C269757C458224B7C895F35F15160DC9@cantor.std.visionutv.se>

Tony, your change fixed the problem!

Thank you so much :)

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 26 september 2012 21:28
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution)

Looking at the code, I can see a solution assuming your theory is correct.
I pushed a patch to HEAD if you care to test it.

On Wed, Sep 26, 2012 at 3:26 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Hello everyone! I experience a strange issue for bridged calls from Lync -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I'm not sure if this a bug or not so I ask the question here first.

The problem is that the audio back from the Asterisk server is not bridged back to the originating client for the first few seconds (how long is different on different calls), instead FS seems to send CN-packets back. I've looked into a wireshark dump, and I see the audio is coming from the Asterisk server (using the correct IP and port), but FS doesn't write the same packet to the other call leg (back to Lync), instead it sends a CN packet every second or so. My guess is that the timestamp is being handled wrong somehow, and FS believes that the real packet is too old (since it has already sent a CN packet with a higher timestamp?), and should not be written to the other leg, then after a while it sends a new CN packet and so on.

The solution is to set  "suppress_cng=true" before bridging the call to the Asterisk server, when this is done, the audio is always bridged correctly, and nothing is missing.

I have pcaps of both working calls (with the variable set) and nonworking calls, so if you believe it might be something that FS should handle differently I'll submit this to Jira.

/Peter

_________________________________________________________________________
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consulting at freeswitch.org<mailto:consulting at freeswitch.org>
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From sunzhimailbox at gmail.com  Thu Sep 27 13:03:01 2012
From: sunzhimailbox at gmail.com (zhi sun)
Date: Thu, 27 Sep 2012 17:03:01 +0800
Subject: [Freeswitch-users] Fwd: [Freeswitch-dev] :how to let the version
 2.1.3 as well as the latest (from git) work for the 0911 and 5001 dialplan?
 there is no problem with version 1.06
In-Reply-To: <CAA0cQJeUsJGBQ-LfwxdwWxUkvR=Ffum0x73VwJd5hCoT1tkcng@mail.gmail.com>
References: <CAA0cQJc_Dm5W6GbOWBW7hMQB_xSM72Kpk2_y2A5dcO6d10gwvw@mail.gmail.com>
	<CC896CDC.543F5%krice@freeswitch.org>
	<CAA0cQJepzETnBCiF01TBXRPy8yf8EgGAEkyn1P5A1UmhRP8rkA@mail.gmail.com>
	<CAA0cQJeUsJGBQ-LfwxdwWxUkvR=Ffum0x73VwJd5hCoT1tkcng@mail.gmail.com>
Message-ID: <CAA0cQJfQQSjAy2MVZYK9zgp7EixYc2X-E2N=aczn+tuVKC=+Ag@mail.gmail.com>

just forward this email to users group, in case anyone else have
encountered the same problem!

thanks for your help
-zhisun


---------- Forwarded message ----------
From: zhi sun <sunzhimailbox at gmail.com>
Date: 2012/9/27
Subject: Re: [Freeswitch-dev] :how to let the version 2.1.3 as well as the
latest (from git) work for the 0911 and 5001 dialplan? there is no problem
with version 1.06
To: freeswitch-dev at lists.freeswitch.org


what i am concerning is how to make the version 1.2.3, as well as the
latest git version works for 0911 and 5001 in default.xml, although the
version 1.0.6 only works partly on this issue.

this problem is very easy to reproduce!!!!


2012/9/27 zhi sun <sunzhimailbox at gmail.com>

> thank ken for your response.
>
> how to update the configs? do you mean the conf folder? i am new to
> freeswitch.
>
> what i have done is just use the conf created by make install command.
>
> since i set different installation path by using ./configure
> --prefix=/my/path, either ver 1.2.3 or ver 1.0.6 has different conf folder
> i think.
>
> at least, i believe that the config with the version 1.2.3 is shipped with
> ver 1.2.3.
>
> thanks,
> -zhisun
>
>
> 2012/9/27 Ken Rice <krice at freeswitch.org>
>
>>  Did you update your configs? 1.0.6 is so old and has more issues then I
>> can count... And configs have been update to resolve problems...
>>
>>
>> On 9/27/12 2:02 AM, "zhi sun" <sunzhimailbox at gmail.com> wrote:
>>
>> further testing results:
>>
>> - for version 1.0.6, the 0911 works, but 5001 doesn't.
>>
>> - for version 1.2.3 and 1.3, both 0911 and 5001 doesn't work.
>>
>> the following are logs for 0911, version 1.2.3
>>
>> the 1000,1001,1002,1003 should be called!!!!
>>
>> ==============================================
>> 2012-09-27 14:58:47.233466 [NOTICE] switch_channel.c:951 New Channel
>> sofia/internal/1001 at 192.168.0.100 [c8c8f364-0870-11e2-bd7f-df6c04573b9b]
>> 2012-09-27 14:58:47.273467 [INFO] mod_dialplan_xml.c:485 Processing 1001
>> <1001>->0911 in context default
>> 2012-09-27 14:58:47.293472 [INFO] switch_core_session.c:2392 Sending
>> early media
>> 2012-09-27 14:58:47.293472 [NOTICE] sofia_glue.c:4226 Pre-Answer
>> sofia/internal/1001 at 192.168.0.100!
>> 2012-09-27 14:58:47.293472 [NOTICE] mod_conference.c:7211 Channel [
>> sofia/internal/1001 at 192.168.0.100] has been answered
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8315 using channel
>> sound prefix:
>> /home/sunzhi/sunzhi/freeswitch/freeswitch-1.2.3/rel/sounds/en/us/callie
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '0' to 'mute'
>> 2012-09-27 14:58:47.293472 [INFO] switch_ivr_async.c:194 Digit parser
>> mod_conference: Setting realm to 'conf'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '*' to 'deaf mute'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '9' to 'energy up'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '8' to 'energy equ'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '7' to 'energy dn'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '3' to 'vol talk up'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '2' to 'vol talk zero'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '1' to 'vol talk dn'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '6' to 'vol listen up'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '5' to 'vol listen zero'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '4' to 'vol listen dn'
>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>> sofia/internal/1001 at 192.168.0.100 binding '#' to 'hangup'
>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1001 at 192.168.0.100]error
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1000 at 192.168.0.100]sofia
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1002 at 192.168.0.100]sofia
>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1002 at 192.168.0.100]sofia]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1001 at 192.168.0.100]error]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1003 at 192.168.0.100]error
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1003 at 192.168.0.100]error]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1000 at 192.168.0.100]sofia]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1004 at 192.168.0.100]error
>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1004 at 192.168.0.100]error]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> ========================================================
>>
>>
>> 2012/9/27 zhi sun <sunzhimailbox at gmail.com>
>>
>> in addition to previous email, i also try the following:
>>
>> - add a new diaplan (5002) similar to 5001 in default.xml
>>
>>     <extension name="dynamic_conference">
>>       <condition field="destination_number" expression="^5002$">
>>     <action application="conference" data="bridge:mydynaconf:
>> sofia/${use_profile}/1002 at 192.168.0.100"/>
>>       </condition>
>>     </extension>
>>
>> - reloadxml
>>
>> - then call 5002 from 1010 sip client,
>>
>> - the same problem happens
>>
>> PS: it works fine on version 1.06
>>
>> ==========================================================
>> freeswitch at mydev.mydomain.com> reloadxml
>>
>> +OK [Success]
>>
>> 2012-09-27 09:46:46.613468 [INFO] mod_enum.c:871 ENUM Reloaded
>> 2012-09-27 09:46:46.613468 [INFO] switch_time.c:1163 Timezone reloaded
>> 530 definitions
>> freeswitch at mydev.mydomain.com> 2012-09-27 09:47:19.413468 [NOTICE]
>> switch_channel.c:951 New Channel sofia/internal/1010 at 192.168.0.100[45f8ccbe-0845-11e2-b58c-8d63071eb0f5]
>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_NEW
>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_state_machine.c:416 (
>> sofia/internal/1010 at 192.168.0.100) State NEW
>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.433469 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>> by acl "domains". Falling back to Digest auth.
>> 2012-09-27 09:47:19.433469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.433469 [DEBUG] sofia.c:1728 detaching session
>> 45f8ccbe-0845-11e2-b58c-8d63071eb0f5
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:1820 Re-attaching to session
>> 45f8ccbe-0845-11e2-b58c-8d63071eb0f5
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>> by acl "domains". Falling back to Digest auth.
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1010 at 192.168.0.100 entering state [received][100]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6293 Remote SDP:
>> v=0
>> o=1010 3519 3519 IN IP4 192.168.0.10
>> s=Talk
>> c=IN IP4 192.168.0.10
>> t=0 0
>> m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
>> a=rtpmap:112 speex/32000
>> a=fmtp:112 vbr=on
>> a=rtpmap:111 speex/16000
>> a=fmtp:111 vbr=on
>> a=rtpmap:110 speex/8000
>> a=fmtp:110 vbr=on
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-11
>>
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6506 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/1010 at 192.168.0.100) State INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:86
>> sofia/internal/1010 at 192.168.0.100 SOFIA INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:126 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_INIT -> CS_ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/1010 at 192.168.0.100) State INIT going to sleep
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1964 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change DOWN -> RINGING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/1010 at 192.168.0.100) State ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:149
>> sofia/internal/1010 at 192.168.0.100 SOFIA ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:117
>> sofia/internal/1010 at 192.168.0.100 Standard ROUTING
>> 2012-09-27 09:47:19.453469 [INFO] mod_dialplan_xml.c:485 Processing 1010
>> <1010>->5002 in context default
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->unloop]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->tod_example] continue=true
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/Time Match (PASS)
>> [tod_example] break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action set(open=true)
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->holiday_example] continue=true
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/TimeMatch (FAIL)
>> [holiday_example] break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->global-intercept] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [global-intercept] destination_number(5002) =~ /^886$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group-intercept] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group-intercept] destination_number(5002) =~ /^\*8$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->intercept-ext] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [intercept-ext]
>> destination_number(5002) =~ /^\*\*(\d+)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->redial]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [redial]
>> destination_number(5002) =~ /^(redial|870)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->global]
>> continue=true
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>> ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>> ${sip_has_crypto}() =~
>> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Absolute Condition [global]
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> hash(insert/${domain_name}-last_dial/global/${uuid})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->snom-demo-2] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-2]
>> destination_number(5002) =~ /^9001$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->snom-demo-1] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-1]
>> destination_number(5002) =~ /^9000$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->eavesdrop]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>> destination_number(5002) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->eavesdrop]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>> destination_number(5002) =~ /^779$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->call_return] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [call_return]
>> destination_number(5002) =~ /^\*69$|^869$|^lcr$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->del-group]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [del-group]
>> destination_number(5002) =~ /^80(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->add-group]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [add-group]
>> destination_number(5002) =~ /^81(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->call-group-simo] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [call-group-simo] destination_number(5002) =~ /^82(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->call-group-order] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [call-group-order] destination_number(5002) =~ /^83(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->extension-intercom] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [extension-intercom] destination_number(5002) =~ /^8(10[01][0-9])$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->Local_Extension] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [Local_Extension] destination_number(5002) =~ /^(10[01][0-9])$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->Local_Extension_Skinny] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [Local_Extension_Skinny] destination_number(5002) =~ /^(11[01][0-9])$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group_dial_sales] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group_dial_sales] destination_number(5002) =~ /^2000$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group_dial_support] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group_dial_support] destination_number(5002) =~ /^2001$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group_dial_billing] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group_dial_billing] destination_number(5002) =~ /^2002$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->operator]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [operator]
>> destination_number(5002) =~ /^(operator|0)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->vmain]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [vmain]
>> destination_number(5002) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->sip_uri]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [sip_uri]
>> destination_number(5002) =~ /^sip:(.*)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->nb_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [nb_conferences] destination_number(5002) =~ /^(30\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->wb_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [wb_conferences] destination_number(5002) =~ /^(31\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->uwb_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [uwb_conferences] destination_number(5002) =~ /^(32\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->cdquality_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [cdquality_conferences] destination_number(5002) =~ /^(33\d{2})$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->freeswitch_public_conf_via_sip] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [freeswitch_public_conf_via_sip] destination_number(5002) =~
>> /^9(888|8888|1616|3232)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->mad_boss_intercom] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [mad_boss_intercom] destination_number(5002) =~ /^0911$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->mad_boss_intercom] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [mad_boss_intercom] destination_number(5002) =~ /^0912$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->mad_boss]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [mad_boss]
>> destination_number(5002) =~ /^0913$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->ivr_demo]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [ivr_demo]
>> destination_number(5002) =~ /^5000$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->dynamic_conference] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [dynamic_conference] destination_number(5002) =~ /^5001$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->dynamic_conference] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS)
>> [dynamic_conference] destination_number(5002) =~ /^5002$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> conference(bridge:mydynaconf:sofia/${use_profile}/1002 at 192.168.0.100)
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:167 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_ROUTING -> CS_EXECUTE
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/1010 at 192.168.0.100) State ROUTING going to sleep
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_EXECUTE
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:453 (
>> sofia/internal/1010 at 192.168.0.100) State EXECUTE
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:242
>> sofia/internal/1010 at 192.168.0.100 SOFIA EXECUTE
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:209
>> sofia/internal/1010 at 192.168.0.100 Standard EXECUTE
>> EXECUTE sofia/internal/1010 at 192.168.0.100 set(open=true)
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET [open]=[true]
>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-spymap/1010/45f8ccbe-0845-11e2-b58c-8d63071eb0f5)
>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/1010/5002)
>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/global/45f8ccbe-0845-11e2-b58c-8d63071eb0f5)
>> EXECUTE sofia/internal/1010 at 192.168.0.100 export(RFC2822_DATE=Thu, 27
>> Sep 2012 09:47:19 +0800)
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1118 EXPORT
>> (export_vars) [RFC2822_DATE]=[Thu, 27 Sep 2012 09:47:19 +0800]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:2390 Application
>> conference Requires media! pre_answering channel
>> sofia/internal/1010 at 192.168.0.100
>> 2012-09-27 09:47:19.453469 [INFO] switch_core_session.c:2392 Sending
>> early media
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[G722:9:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[G722:9:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[G722:9:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[G722:9:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3077 Set Codec
>> sofia/internal/1010 at 192.168.0.100 GSM/8000 20 ms 160 samples 13200 bits
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_codec.c:111
>> sofia/internal/1010 at 192.168.0.100 Original read codec set to GSM:3
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5219 Set 2833 dtmf
>> send/recv payload to 101
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3327 AUDIO RTP [
>> sofia/internal/1010 at 192.168.0.100] 192.168.0.100 port 31092 ->
>> 192.168.0.10 port 7078 codec: 3 ms: 20
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_rtp.c:1927 Starting timer
>> [soft] 160 bytes per 20ms
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3591 Set 2833 dtmf send
>> payload to 101
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3597 Set 2833 dtmf
>> receive payload to 101
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3624
>> sofia/internal/1010 at 192.168.0.100 Set rtp dtmf delay to 40
>> 2012-09-27 09:47:19.453469 [NOTICE] sofia_glue.c:4226 Pre-Answer
>> sofia/internal/1010 at 192.168.0.100!
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:3092 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change RINGING -> EARLY
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:2730 Ring SDP:
>> v=0
>> o=FreeSWITCH 1348679347 1348679348 IN IP4 192.168.0.100
>> s=FreeSWITCH
>> c=IN IP4 192.168.0.100
>> t=0 0
>> m=audio 31092 RTP/AVP 3 101
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:759 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> EXECUTE sofia/internal/1010 at 192.168.0.100 conference(bridge:mydynaconf:
>> sofia/internal/1002 at 192.168.0.100)
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1010 at 192.168.0.100 entering state [early][183]
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:836 Local SDP
>> sofia/internal/1010 at 192.168.0.100:
>> v=0
>> o=FreeSWITCH 1348679347 1348679349 IN IP4 192.168.0.100
>> s=FreeSWITCH
>> c=IN IP4 192.168.0.100
>> t=0 0
>> m=audio 31092 RTP/AVP 3 101
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:759 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:3351 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change EARLY -> ACTIVE
>> 2012-09-27 09:47:19.453469 [NOTICE] mod_conference.c:7211 Channel [
>> sofia/internal/1010 at 192.168.0.100] has been answered
>> 2012-09-27 09:47:19.453469 [INFO] mod_conference.c:8315 using channel
>> sound prefix:
>> /home/sunzhi/sunzhi/freeswitch/freeswitch-1.2.3/rel/sounds/en/us/callie
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_conference.c:1922 Setup timer
>> success interval: 20  samples: 160
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1072
>> sofia/internal/1010 at 192.168.0.100 EXPORTING[export_vars]
>> [RFC2822_DATE]=[Thu, 27 Sep 2012 09:47:19 +0800] to event
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:47:19.453469 [NOTICE] switch_channel.c:951 New Channel
>> sofia/internal/1002 at 192.168.0.100 [46000434-0845-11e2-b598-8d63071eb0f5]
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:4879 (
>> sofia/internal/1002 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:4954 [zrtp_passthru]
>> Setting a-leg inherit_codec=true
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:4957 [zrtp_passthru]
>> Setting b-leg absolute_codec_string='GSM at 8000h@20i at 13200b,PCMU at 8000h
>> @20i at 64000b,PCMA at 8000h@20i at 64000b'
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1010 at 192.168.0.100 entering state [completed][200]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/1002 at 192.168.0.100) State INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:86
>> sofia/internal/1002 at 192.168.0.100 SOFIA INIT
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:2637 Local SDP:
>> v=0
>> o=FreeSWITCH 1348687005 1348687006 IN IP4 192.168.0.100
>> s=FreeSWITCH
>> c=IN IP4 192.168.0.100
>> t=0 0
>> m=audio 23434 RTP/AVP 3 0 8 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:126 (
>> sofia/internal/1002 at 192.168.0.100) State Change CS_INIT -> CS_ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/1002 at 192.168.0.100) State INIT going to sleep
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1964 (
>> sofia/internal/1002 at 192.168.0.100) Callstate Change DOWN -> RINGING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/1002 at 192.168.0.100) State ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:149
>> sofia/internal/1002 at 192.168.0.100 SOFIA ROUTING
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_ivr_originate.c:67 (
>> sofia/internal/1002 at 192.168.0.100) State Change CS_ROUTING ->
>> CS_CONSUME_MEDIA
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/1002 at 192.168.0.100) State ROUTING going to sleep
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_CONSUME_MEDIA
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:465 (
>> sofia/internal/1002 at 192.168.0.100) State CONSUME_MEDIA
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:465 (
>> sofia/internal/1002 at 192.168.0.100) State CONSUME_MEDIA going to sleep
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1002 at 192.168.0.100 entering state [calling][0]
>> 2012-09-27 09:47:19.453469 [NOTICE] switch_channel.c:951 New Channel
>> sofia/internal/0000000000 at 192.168.0.100[46006712-0845-11e2-b59c-8d63071eb0f5]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_NEW
>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:416 (
>> sofia/internal/0000000000 at 192.168.0.100) State NEW
>> 2012-09-27 09:47:19.473478 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:8412 IP 192.168.0.100 Rejected
>> by acl "domains". Falling back to Digest auth.
>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/0000000000 at 192.168.0.100 entering state [received][100]
>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:6293 Remote SDP:
>> v=0
>> o=FreeSWITCH 1348687005 1348687006 IN IP4 192.168.0.100
>> s=FreeSWITCH
>> c=IN IP4 192.168.0.100
>> t=0 0
>> m=audio 23434 RTP/AVP 3 0 8 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:6506 (
>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_INIT
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/0000000000 at 192.168.0.100) State INIT
>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:86
>> sofia/internal/0000000000 at 192.168.0.100 SOFIA INIT
>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:126 (
>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_INIT ->
>> CS_ROUTING
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/0000000000 at 192.168.0.100) State INIT going to sleep
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_ROUTING
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_channel.c:1964 (
>> sofia/internal/0000000000 at 192.168.0.100) Callstate Change DOWN -> RINGING
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/0000000000 at 192.168.0.100) State ROUTING
>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:149
>> sofia/internal/0000000000 at 192.168.0.100 SOFIA ROUTING
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:117
>> sofia/internal/0000000000 at 192.168.0.100 Standard ROUTING
>> 2012-09-27 09:47:19.493470 [INFO] mod_dialplan_xml.c:485 Processing
>> FreeSWITCH <0000000000>->1002 in context public
>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 parsing
>> [public->unloop] continue=false
>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 Regex (PASS) [unloop]
>> ${sip_looped_call}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 Action
>> deflect(${destination_number})
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:167 (
>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/0000000000 at 192.168.0.100) State ROUTING going to sleep
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_EXECUTE
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:453 (
>> sofia/internal/0000000000 at 192.168.0.100) State EXECUTE
>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:242
>> sofia/internal/0000000000 at 192.168.0.100 SOFIA EXECUTE
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:209
>> sofia/internal/0000000000 at 192.168.0.100 Standard EXECUTE
>> EXECUTE sofia/internal/0000000000 at 192.168.0.100 deflect(1002)
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:7308 Process REFER to [
>> 1002 at 192.168.0.100]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_ivr.c:1742 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_EXECUTE -> CS_ROUTING
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:759 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [NOTICE] switch_ivr.c:1748 Transfer
>> sofia/internal/1010 at 192.168.0.100 to XML[1002 at default]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2950 (
>> sofia/internal/1002 at 192.168.0.100) Callstate Change RINGING -> HANGUP
>> 2012-09-27 09:47:19.513472 [NOTICE] switch_ivr_originate.c:3326 Hangup
>> sofia/internal/1002 at 192.168.0.100 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2973 Send signal
>> sofia/internal/1002 at 192.168.0.100 [KILL]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_HANGUP
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_ivr_originate.c:3502 Originate
>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>> 2012-09-27 09:47:19.513472 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: ORIGINATOR_CANCEL
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2950 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change ACTIVE -> HANGUP
>> 2012-09-27 09:47:19.513472 [NOTICE] mod_conference.c:6617 Hangup
>> sofia/internal/1010 at 192.168.0.100 [CS_ROUTING] [ORIGINATOR_CANCEL]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/1002 at 192.168.0.100) State HANGUP
>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:483 Channel
>> sofia/internal/1002 at 192.168.0.100 hanging up, cause: ORIGINATOR_CANCEL
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2973 Send signal
>> sofia/internal/1010 at 192.168.0.100 [KILL]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:2553
>> sofia/internal/1010 at 192.168.0.100 skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:453 (
>> sofia/internal/1010 at 192.168.0.100) State EXECUTE going to sleep
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_HANGUP
>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:542 Sending CANCEL to
>> sofia/internal/1002 at 192.168.0.100
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:48
>> sofia/internal/1002 at 192.168.0.100 Standard HANGUP, cause:
>> ORIGINATOR_CANCEL
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/1002 at 192.168.0.100) State HANGUP going to sleep
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:429 (
>> sofia/internal/1002 at 192.168.0.100) State Change CS_HANGUP -> CS_REPORTING
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_REPORTING
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/1002 at 192.168.0.100) State REPORTING
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:92
>> sofia/internal/1002 at 192.168.0.100 Standard REPORTING, cause:
>> ORIGINATOR_CANCEL
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/1002 at 192.168.0.100) State REPORTING going to sleep
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:423 (
>> sofia/internal/1002 at 192.168.0.100) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/1010 at 192.168.0.100) State HANGUP
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1415 Session 5 (
>> sofia/internal/1002 at 192.168.0.100) Locked, Waiting on external entities
>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:483 Channel
>> sofia/internal/1010 at 192.168.0.100 hanging up, cause: ORIGINATOR_CANCEL
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:532 Sending BYE to
>> sofia/internal/1010 at 192.168.0.100
>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1433 Session 5 (
>> sofia/internal/1002 at 192.168.0.100) Ended
>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1437 Close
>> Channel sofia/internal/1002 at 192.168.0.100 [CS_DESTROY]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:48
>> sofia/internal/1010 at 192.168.0.100 Standard HANGUP, cause:
>> ORIGINATOR_CANCEL
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/1010 at 192.168.0.100) State HANGUP going to sleep
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:429 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_HANGUP -> CS_REPORTING
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_REPORTING
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/1010 at 192.168.0.100) State REPORTING
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:527 (
>> sofia/internal/1002 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:92
>> sofia/internal/1010 at 192.168.0.100 Standard REPORTING, cause:
>> ORIGINATOR_CANCEL
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/1010 at 192.168.0.100) State REPORTING going to sleep
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:530 (
>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/1002 at 192.168.0.100) State DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:376
>> sofia/internal/1002 at 192.168.0.100 SOFIA DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:99
>> sofia/internal/1002 at 192.168.0.100 Standard DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/1002 at 192.168.0.100) State DESTROY going to sleep
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:423 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1415 Session 4 (
>> sofia/internal/1010 at 192.168.0.100) Locked, Waiting on external entities
>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1433 Session 4 (
>> sofia/internal/1010 at 192.168.0.100) Ended
>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1437 Close
>> Channel sofia/internal/1010 at 192.168.0.100 [CS_DESTROY]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:527 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:530 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/1010 at 192.168.0.100) State DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:376
>> sofia/internal/1010 at 192.168.0.100 SOFIA DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:99
>> sofia/internal/1010 at 192.168.0.100 Standard DESTROY
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/1010 at 192.168.0.100) State DESTROY going to sleep
>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.533467 [DEBUG] mod_conference.c:2461 Write Lock ON
>> 2012-09-27 09:47:19.533467 [DEBUG] mod_conference.c:2464 Write Lock OFF
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_channel.c:2950 (
>> sofia/internal/0000000000 at 192.168.0.100) Callstate Change RINGING ->
>> HANGUP
>> 2012-09-27 09:47:19.593467 [NOTICE] sofia.c:420 Hangup
>> sofia/internal/0000000000 at 192.168.0.100 [CS_EXECUTE] [BLIND_TRANSFER]
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_channel.c:2973 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [KILL]
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:2553
>> sofia/internal/0000000000 at 192.168.0.100 skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:453 (
>> sofia/internal/0000000000 at 192.168.0.100) State EXECUTE going to sleep
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_HANGUP
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/0000000000 at 192.168.0.100) State HANGUP
>> 2012-09-27 09:47:19.593467 [DEBUG] mod_sofia.c:483 Channel
>> sofia/internal/0000000000 at 192.168.0.100 hanging up, cause: BLIND_TRANSFER
>> 2012-09-27 09:47:19.593467 [DEBUG] mod_sofia.c:613 Responding to INVITE
>> with: 480
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:48
>> sofia/internal/0000000000 at 192.168.0.100 Standard HANGUP, cause:
>> BLIND_TRANSFER
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/0000000000 at 192.168.0.100) State HANGUP going to sleep
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:429 (
>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/0000000000 at 192.168.0.100) Running State Change
>> CS_REPORTING
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/0000000000 at 192.168.0.100) State REPORTING
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:92
>> sofia/internal/0000000000 at 192.168.0.100 Standard REPORTING, cause:
>> BLIND_TRANSFER
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/0000000000 at 192.168.0.100) State REPORTING going to sleep
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:423 (
>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1415 Session 6 (
>> sofia/internal/0000000000 at 192.168.0.100) Locked, Waiting on external
>> entities
>> 2012-09-27 09:47:19.593467 [NOTICE] switch_core_session.c:1433 Session 6 (
>> sofia/internal/0000000000 at 192.168.0.100) Ended
>> 2012-09-27 09:47:19.593467 [NOTICE] switch_core_session.c:1437 Close
>> Channel sofia/internal/0000000000 at 192.168.0.100 [CS_DESTROY]
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:527 (
>> sofia/internal/0000000000 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:530 (
>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_DESTROY
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/0000000000 at 192.168.0.100) State DESTROY
>> 2012-09-27 09:47:19.593467 [DEBUG] mod_sofia.c:376
>> sofia/internal/0000000000 at 192.168.0.100 SOFIA DESTROY
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:99
>> sofia/internal/0000000000 at 192.168.0.100 Standard DESTROY
>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/0000000000 at 192.168.0.100) State DESTROY going to sleep
>>
>> ============================================================================
>>
>>
>>
>>
>> 2012/9/27 zhi sun <sunzhimailbox at gmail.com>
>>
>> thanks for response, the detailed debug log as below:
>>
>> version 2.1.3
>>
>> there are two sip client: 1002, 1010
>>
>> i can call 1002 from 1010 successfully.
>>
>> then i try to call 0911 from 1010, as you know the 1002 is a member of
>> sales group, the 0911 will try to out call 1002 according the rule in
>> default.xml
>>
>> =============================================
>> 2012-09-27 09:34:07.533467 [NOTICE] switch_channel.c:951 New Channel
>> sofia/internal/1010 at 192.168.0.100 [6dfa8876-0843-11e2-b574-8d63071eb0f5]
>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_NEW
>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_state_machine.c:416 (
>> sofia/internal/1010 at 192.168.0.100) State NEW
>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.553467 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>> by acl "domains". Falling back to Digest auth.
>> 2012-09-27 09:34:07.553467 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.553467 [DEBUG] sofia.c:1728 detaching session
>> 6dfa8876-0843-11e2-b574-8d63071eb0f5
>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:1820 Re-attaching to session
>> 6dfa8876-0843-11e2-b574-8d63071eb0f5
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>> by acl "domains". Falling back to Digest auth.
>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1010 at 192.168.0.100 entering state [received][100]
>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:6293 Remote SDP:
>> v=0
>> o=1010 3118 3118 IN IP4 192.168.0.10
>> s=Talk
>> c=IN IP4 192.168.0.10
>> t=0 0
>> m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
>> a=rtpmap:112 speex/32000
>> a=fmtp:112 vbr=on
>> a=rtpmap:111 speex/16000
>> a=fmtp:111 vbr=on
>> a=rtpmap:110 speex/8000
>> a=fmtp:110 vbr=on
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-11
>>
>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:6506 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_INIT
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/1010 at 192.168.0.100) State INIT
>> 2012-09-27 09:34:07.573471 [DEBUG] mod_sofia.c:86
>> sofia/internal/1010 at 192.168.0.100 SOFIA INIT
>> 2012-09-27 09:34:07.573471 [DEBUG] mod_sofia.c:126 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_INIT -> CS_ROUTING
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:437 (
>> sofia/internal/1010 at 192.168.0.100) State INIT going to sleep
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_ROUTING
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_channel.c:1964 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change DOWN -> RINGING
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/1010 at 192.168.0.100) State ROUTING
>> 2012-09-27 09:34:07.573471 [DEBUG] mod_sofia.c:149
>> sofia/internal/1010 at 192.168.0.100 SOFIA ROUTING
>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:117
>> sofia/internal/1010 at 192.168.0.100 Standard ROUTING
>> 2012-09-27 09:34:07.573471 [INFO] mod_dialplan_xml.c:485 Processing 1010
>> <1010>->0911 in context default
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->unloop]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->tod_example] continue=true
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/Time Match (PASS)
>> [tod_example] break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action set(open=true)
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->holiday_example] continue=true
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/TimeMatch (FAIL)
>> [holiday_example] break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->global-intercept] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [global-intercept] destination_number(0911) =~ /^886$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group-intercept] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group-intercept] destination_number(0911) =~ /^\*8$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->intercept-ext] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [intercept-ext]
>> destination_number(0911) =~ /^\*\*(\d+)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->redial]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [redial]
>> destination_number(0911) =~ /^(redial|870)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->global]
>> continue=true
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>> ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>> ${sip_has_crypto}() =~
>> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Absolute Condition [global]
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> hash(insert/${domain_name}-last_dial/global/${uuid})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->snom-demo-2] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-2]
>> destination_number(0911) =~ /^9001$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->snom-demo-1] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-1]
>> destination_number(0911) =~ /^9000$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->eavesdrop]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>> destination_number(0911) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->eavesdrop]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>> destination_number(0911) =~ /^779$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->call_return] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [call_return]
>> destination_number(0911) =~ /^\*69$|^869$|^lcr$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->del-group]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [del-group]
>> destination_number(0911) =~ /^80(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->add-group]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [add-group]
>> destination_number(0911) =~ /^81(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->call-group-simo] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [call-group-simo] destination_number(0911) =~ /^82(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->call-group-order] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [call-group-order] destination_number(0911) =~ /^83(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->extension-intercom] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [extension-intercom] destination_number(0911) =~ /^8(10[01][0-9])$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->Local_Extension] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [Local_Extension] destination_number(0911) =~ /^(10[01][0-9])$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->Local_Extension_Skinny] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [Local_Extension_Skinny] destination_number(0911) =~ /^(11[01][0-9])$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group_dial_sales] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group_dial_sales] destination_number(0911) =~ /^2000$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group_dial_support] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group_dial_support] destination_number(0911) =~ /^2001$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->group_dial_billing] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [group_dial_billing] destination_number(0911) =~ /^2002$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->operator]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [operator]
>> destination_number(0911) =~ /^(operator|0)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->vmain]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [vmain]
>> destination_number(0911) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->sip_uri]
>> continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [sip_uri]
>> destination_number(0911) =~ /^sip:(.*)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->nb_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [nb_conferences] destination_number(0911) =~ /^(30\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->wb_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [wb_conferences] destination_number(0911) =~ /^(31\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->uwb_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [uwb_conferences] destination_number(0911) =~ /^(32\d{2})$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->cdquality_conferences] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [cdquality_conferences] destination_number(0911) =~ /^(33\d{2})$/
>> break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->freeswitch_public_conf_via_sip] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>> [freeswitch_public_conf_via_sip] destination_number(0911) =~
>> /^9(888|8888|1616|3232)$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>> [default->mad_boss_intercom] continue=false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS)
>> [mad_boss_intercom] destination_number(0911) =~ /^0911$/ break=on-false
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> set(conference_auto_outcall_caller_id_name=Mad Boss1)
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> set(conference_auto_outcall_caller_id_number=0911)
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> set(conference_auto_outcall_timeout=60)
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> set(conference_auto_outcall_flags=mute)
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> set(conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
>> 2 a s1 transfer::intercept:${uuid} inline'})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> set(sip_exclude_contact=${network_addr})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> conference_set_auto_outcall(${group_call(sales)})
>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>> conference(madboss_intercom1 at default+flags{endconf|deaf})
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:167 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_ROUTING -> CS_EXECUTE
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:446 (
>> sofia/internal/1010 at 192.168.0.100) State ROUTING going to sleep
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_EXECUTE
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:453 (
>> sofia/internal/1010 at 192.168.0.100) State EXECUTE
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_sofia.c:242
>> sofia/internal/1010 at 192.168.0.100 SOFIA EXECUTE
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:209
>> sofia/internal/1010 at 192.168.0.100 Standard EXECUTE
>> EXECUTE sofia/internal/1010 at 192.168.0.100 set(open=true)
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET [open]=[true]
>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-spymap/1010/6dfa8876-0843-11e2-b574-8d63071eb0f5)
>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/1010/0911)
>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/global/6dfa8876-0843-11e2-b574-8d63071eb0f5)
>> EXECUTE sofia/internal/1010 at 192.168.0.100 export(RFC2822_DATE=Thu, 27
>> Sep 2012 09:34:07 +0800)
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_channel.c:1118 EXPORT
>> (export_vars) [RFC2822_DATE]=[Thu, 27 Sep 2012 09:34:07 +0800]
>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_caller_id_name=Mad Boss1)
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET
>> [conference_auto_outcall_caller_id_name]=[Mad Boss1]
>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_caller_id_number=0911)
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET
>> [conference_auto_outcall_caller_id_number]=[0911]
>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_timeout=60)
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET
>> [conference_auto_outcall_timeout]=[60]
>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_flags=mute)
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET
>> [conference_auto_outcall_flags]=[mute]
>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
>> 2 a s1 transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline'})
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET
>> [conference_auto_outcall_prefix]=[{sip_auto_answer=true,execute_on_answer='bind_meta_app
>> 2 a s1 transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline'}]
>> EXECUTE sofia/internal/1010 at 192.168.0.100set(sip_exclude_contact=192.168.0.10)
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>> sofia/internal/1010 at 192.168.0.100 SET
>> [sip_exclude_contact]=[192.168.0.10]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:2390 Application
>> conference_set_auto_outcall Requires media! pre_answering channel
>> sofia/internal/1010 at 192.168.0.100
>> 2012-09-27 09:34:07.593468 [INFO] switch_core_session.c:2392 Sending
>> early media
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[G722:9:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:112:32000:20:0]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[G722:9:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:111:16000:20:0]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[G722:9:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [speex:110:8000:20:0]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[G722:9:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3077 Set Codec
>> sofia/internal/1010 at 192.168.0.100 GSM/8000 20 ms 160 samples 13200 bits
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_codec.c:111
>> sofia/internal/1010 at 192.168.0.100 Original read codec set to GSM:3
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5219 Set 2833 dtmf
>> send/recv payload to 101
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3327 AUDIO RTP [
>> sofia/internal/1010 at 192.168.0.100] 192.168.0.100 port 19426 ->
>> 192.168.0.10 port 7078 codec: 3 ms: 20
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_rtp.c:1927 Starting timer
>> [soft] 160 bytes per 20ms
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3591 Set 2833 dtmf send
>> payload to 101
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3597 Set 2833 dtmf
>> receive payload to 101
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3624
>> sofia/internal/1010 at 192.168.0.100 Set rtp dtmf delay to 40
>> 2012-09-27 09:34:07.593468 [NOTICE] sofia_glue.c:4226 Pre-Answer
>> sofia/internal/1010 at 192.168.0.100!
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_channel.c:3092 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change RINGING -> EARLY
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_sofia.c:2730 Ring SDP:
>> v=0
>> o=FreeSWITCH 1348690221 1348690222 IN IP4 192.168.0.100
>> s=FreeSWITCH
>> c=IN IP4 192.168.0.100
>> t=0 0
>> m=audio 19426 RTP/AVP 3 101
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:759 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> EXECUTE sofia/internal/1010 at 192.168.0.100conference_set_auto_outcall([sip_invite_domain=192.168.0.100,
>> presence_id=1000 at 192.168.0.100]sofia/internal/sip:1000 at 192.168.0.20:5060<
>> http://sip:1000 at 192.168.0.20:5060> ,[sip_invite_domain=192.168.0.100,
>> presence_id=1001 at 192.168.0.100]error/user_not_registered
>> ,[sip_invite_domain=192.168.0.100,
>> presence_id=1002 at 192.168.0.100]error/user_not_registered
>> ,[sip_invite_domain=192.168.0.100,
>> presence_id=1003 at 192.168.0.100]error/user_not_registered
>> ,[sip_invite_domain=192.168.0.100,
>> presence_id=1004 at 192.168.0.100]error/user_not_registered)
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1010 at 192.168.0.100 entering state [early][183]
>> EXECUTE sofia/internal/1010 at 192.168.0.100conference(madboss_intercom1 at default
>> +flags{endconf|deaf})
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_sofia.c:836 Local SDP
>> sofia/internal/1010 at 192.168.0.100:
>> v=0
>> o=FreeSWITCH 1348690221 1348690223 IN IP4 192.168.0.100
>> s=FreeSWITCH
>> c=IN IP4 192.168.0.100
>> t=0 0
>> m=audio 19426 RTP/AVP 3 101
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:759 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_channel.c:3351 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change EARLY -> ACTIVE
>> 2012-09-27 09:34:07.593468 [NOTICE] mod_conference.c:7211 Channel [
>> sofia/internal/1010 at 192.168.0.100] has been answered
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.593468 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1010 at 192.168.0.100 entering state [completed][200]
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8315 using channel
>> sound prefix:
>> /home/sunzhi/sunzhi/freeswitch/freeswitch-1.2.3/rel/sounds/en/us/callie
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:7092 Raw Codec
>> Activation Success L16 at 8000hz 1 channel 20ms
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:7137 Raw Codec
>> Activation Success L16 at 8000hz 1 channel 20ms
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:1922 Setup timer
>> success interval: 20  samples: 160
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_codec.c:219
>> sofia/internal/1010 at 192.168.0.100 Push codec L16:70
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '0' to 'mute'
>> 2012-09-27 09:34:07.593468 [INFO] switch_ivr_async.c:194 Digit parser
>> mod_conference: Setting realm to 'conf'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 0/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bab8
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '*' to 'deaf mute'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding */conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bae8
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '9' to 'energy up'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 9/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bb18
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '8' to 'energy equ'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 8/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bb48
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '7' to 'energy dn'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 7/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bb78
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '3' to 'vol talk up'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 3/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bba8
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '2' to 'vol talk zero'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 2/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bbd8
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '1' to 'vol talk dn'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 1/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bc08
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '6' to 'vol listen up'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 6/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bc38
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '5' to 'vol listen zero'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 5/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bc68
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '4' to 'vol listen dn'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding 4/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bc98
>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>> sofia/internal/1010 at 192.168.0.100 binding '#' to 'hangup'
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>> mod_conference: binding #/conf/0 callback: 0x7f7bcbde4640 data:
>> 0x7f7b8002bcc8
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:759 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:3474 Setup timer soft
>> success interval: 20  samples: 160
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>> session specific variables
>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1000 at 192.168.0.100]sofia
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>> session specific variables
>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1000 at 192.168.0.100]sofia]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1001 at 192.168.0.100]error
>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1001 at 192.168.0.100]error]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>> session specific variables
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1002 at 192.168.0.100]error
>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1002 at 192.168.0.100]error]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>> session specific variables
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>> Thread for outcall
>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>> global variables
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1003 at 192.168.0.100]error
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1003 at 192.168.0.100]error]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [sip_auto_answer]=[true]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>> [execute_on_answer]=[bind_meta_app 2 a s1
>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>> session specific variables
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>> locate channel type presence_id=1004 at 192.168.0.100]error
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>> create outgoing channel of type [presence_id=1004 at 192.168.0.100]error]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> 2012-09-27 09:34:07.613471 [DEBUG] mod_local_stream.c:417 Opening Stream
>> [moh/8000] 8000hz
>> 2012-09-27 09:34:07.613471 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.613471 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.613471 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:07.633467 [DEBUG] sofia.c:6282 Channel
>> sofia/internal/1010 at 192.168.0.100 entering state [ready][200]
>> 2012-09-27 09:34:07.693467 [DEBUG] switch_rtp.c:3596 Correct ip/port
>> confirmed.
>> 2012-09-27 09:34:07.713468 [DEBUG] mod_conference.c:4288 Queueing file
>> 'tone_stream://%(500,0,640)' for play
>> 2012-09-27 09:34:10.733467 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_channel.c:2950 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change ACTIVE -> HANGUP
>> 2012-09-27 09:34:10.753468 [NOTICE] sofia.c:711 Hangup
>> sofia/internal/1010 at 192.168.0.100 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_channel.c:2973 Send signal
>> sofia/internal/1010 at 192.168.0.100 [KILL]
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:10.753468 [DEBUG] mod_conference.c:3777 Channel leaving
>> conference, cause: NORMAL_CLEARING
>> 2012-09-27 09:34:10.753468 [DEBUG] mod_conference.c:7645
>> sofia/internal/1010 at 192.168.0.100 skip receive message [UNBRIDGE]
>> (channel is hungup already)
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_codec.c:244
>> sofia/internal/1010 at 192.168.0.100 Restore previous codec GSM:3.
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:2553
>> sofia/internal/1010 at 192.168.0.100 skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:453 (
>> sofia/internal/1010 at 192.168.0.100) State EXECUTE going to sleep
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_HANGUP
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/1010 at 192.168.0.100) State HANGUP
>> 2012-09-27 09:34:10.753468 [DEBUG] mod_sofia.c:483 Channel
>> sofia/internal/1010 at 192.168.0.100 hanging up, cause: NORMAL_CLEARING
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:48
>> sofia/internal/1010 at 192.168.0.100 Standard HANGUP, cause: NORMAL_CLEARING
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/1010 at 192.168.0.100) State HANGUP going to sleep
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:429 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_HANGUP -> CS_REPORTING
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_REPORTING
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/1010 at 192.168.0.100) State REPORTING
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:92
>> sofia/internal/1010 at 192.168.0.100 Standard REPORTING, cause:
>> NORMAL_CLEARING
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:703 (
>> sofia/internal/1010 at 192.168.0.100) State REPORTING going to sleep
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:423 (
>> sofia/internal/1010 at 192.168.0.100) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1415 Session 3 (
>> sofia/internal/1010 at 192.168.0.100) Locked, Waiting on external entities
>> 2012-09-27 09:34:10.753468 [NOTICE] switch_core_session.c:1433 Session 3 (
>> sofia/internal/1010 at 192.168.0.100) Ended
>> 2012-09-27 09:34:10.753468 [NOTICE] switch_core_session.c:1437 Close
>> Channel sofia/internal/1010 at 192.168.0.100 [CS_DESTROY]
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:527 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:530 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_DESTROY
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/1010 at 192.168.0.100) State DESTROY
>> 2012-09-27 09:34:10.753468 [DEBUG] mod_sofia.c:376
>> sofia/internal/1010 at 192.168.0.100 SOFIA DESTROY
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:99
>> sofia/internal/1010 at 192.168.0.100 Standard DESTROY
>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:540 (
>> sofia/internal/1010 at 192.168.0.100) State DESTROY going to sleep
>> 2012-09-27 09:34:10.773471 [NOTICE] mod_conference.c:2369 Ending pending
>> outcall channels for Conference: 'madboss_intercom1'
>> 2012-09-27 09:34:10.773471 [DEBUG] mod_conference.c:2461 Write Lock ON
>> 2012-09-27 09:34:10.773471 [DEBUG] mod_conference.c:2464 Write Lock OFF
>>
>> ======================================================================
>>
>>
>> 2012/9/27 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> That is not nearly enough log file, you need to supply the entire log of
>> the call in full debug mode
>>
>> On Wed, Sep 26, 2012 at 8:13 AM, zhi sun <sunzhimailbox at gmail.com> wrote:
>>
>> in the default diaplan, the 0911 conference call that make a out call to
>> a group doesn't work for the latest and version 2.1.3. the out call always
>> failed because of CHAN_NOT_IMPLEMENTED,
>>
>> but it works fine for version 1.06
>>
>> the same problem happens with the 5001 in dialplan (default.xml)
>>
>> it is very easy to reproduce this issue: i just get the correct version,
>> make, make install, make cd-sounds-install.
>>
>> the log with problem looks like below:
>>
>> ===========================================
>>
>> 2012-09-26 15:37:10.973469 [NOTICE] switch_ivr.c:1748 Transfer
>> sofia/internal/1000 at 192.168.0.100 to XML[1010 at default]
>> 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>> 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2950 (
>> sofia/internal/1010 at 192.168.0.100) Callstate Change RINGING -> HANGUP
>> 2012-09-26 15:37:10.993469 [NOTICE] switch_ivr_originate.c:3330 Hangup
>> sofia/internal/1010 at 192.168.0.100 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2973 Send signal
>> sofia/internal/1010 at 192.168.0.100 [KILL]
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_ivr_originate.c:3506 Originate
>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_state_machine.c:398 (
>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_HANGUP
>> 2012-09-26 15:37:10.993469 [ERR] mod_conference.c:6626 Cannot create
>> outgoing channel, cause: ORIGINATOR_CANCEL
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2950 (
>> sofia/internal/1000 at 192.168.0.100) Callstate Change ACTIVE -> HANGUP
>> 2012-09-26 15:37:10.993469 [NOTICE] mod_conference.c:6629 Hangup
>> sofia/internal/1000 at 192.168.0.100 [CS_ROUTING] [ORIGINATOR_CANCEL]
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2973 Send signal
>> sofia/internal/1000 at 192.168.0.100 [KILL]
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_state_machine.c:638 (
>> sofia/internal/1010 at 192.168.0.100) State HANGUP
>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_session.c:1210 Send signal
>> sofia/internal/1000 at 192.168.0.100 [BREAK]
>>
>> ============================================
>>
>> i am new to freeswitch, Is there anything i missed to let the version
>> 2.1.3 as well as the latest (from git) work for the 0911 and 5001 dialplan?
>> both of them make an out call from a conference.
>>
>> thanks
>> -zhisun
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> 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
>>
>>
>>
>>
>> --
>> 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-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 vitaliy.davudov at vts24.ru  Thu Sep 27 16:31:30 2012
From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=)
Date: Thu, 27 Sep 2012 16:31:30 +0400
Subject: [Freeswitch-users] Freeswitch HA issue with failover to second node
Message-ID: <50644722.3080904@vts24.ru>

Hi, list!

I've installed Freeswitch HA cluster for failover to second node. It use 
corosync and pacemaker to failover IP-address and Mysql Master-Master 
replication.
After rebooting the first node, the second node successfully recover 
calls, but appears problem with one-way media. Channel with PSTN-gateway 
(throught external profile) is established in 2-3 secs. Channel with 
registered user (throught internal profile) is established only after 
~30 secs! Internal and external profiles listening to the same 
ip-address. I think sip-packets blocked by FS acl. When I do tcpdump, 
packets from FS arrive to the user ip-phone, but sip-packets from 
ip-phone didn't arrive to FS.

This is from user directory:

           <user id="111" cidr="172.30.0.11/32">
             <params>
               <param name="password" value="bla-bla-bla"/>
             </params>
             <variables>
               <variable name="user_context" 
value="private-vip-telecom-service"/>
               <variable name="effective_caller_id_number" 
value="4959894700"/>
               <variable name="outbound_caller_id_number" 
value="4959894700"/>
               <variable name="callgroup" value="support"/>
             </variables>
           </user>

In sip-profile named "internal" enabled only one acl:

<param name="apply-inbound-acl" value="domains"/>

In acl.conf.xml:

     <list name="domains" default="deny">
       <!-- domain= is special it scans the domain from the directory to 
build the ACL
       <node type="allow" domain="$${domain}"/> -->
       <node type="allow" domain="172.30.0.1"/>
       <!-- use cidr= if you wish to allow ip ranges to this domains 
acl. -->
       <!-- <node type="allow" cidr="192.168.0.0/24"/> -->
     </list>

And from FS console:

freeswitch at internal> acl 172.30.0.11 domains
true

Here you can see FS console log from second node after sip recover: 
http://pastebin.freeswitch.org/19941

If anybody has ideas how to resolve this issue, I'll be very happy!

-- 
Best regards,
Vitaly.



From ntomer at newgen.co.in  Thu Sep 27 16:32:28 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 27 Sep 2012 18:02:28 +0530
Subject: [Freeswitch-users] Help needed in a Contact Center solution
Message-ID: <011601cd9cac$2919d840$7b4d88c0$@co.in>

Hi,

 

I have posted this query earlier, now refining it a bit more. I need to do
the following using FreeSwithc -


1. An end-customer calls, the call is handled by mod_ivr.
2. Customer is presented with a menu, she makes selection and also enters
some identification. Her call is parked on a pre-decided extension and an
entry is made in a database for this call. This entry will hold the
selections made by the caller and the extension where call is parked.
3. I will write a service which will poll this database for new calls.

5. New calls will be shown in agents' dashboards.
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 points 1 & 2. 



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 gabe at gundy.org  Thu Sep 27 16:53:25 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 27 Sep 2012 06:53:25 -0600
Subject: [Freeswitch-users] Help needed in a Contact Center solution
In-Reply-To: <011601cd9cac$2919d840$7b4d88c0$@co.in>
References: <011601cd9cac$2919d840$7b4d88c0$@co.in>
Message-ID: <CAMwZ-tJnaHnwq1P+Z91O+_vz54xmPjdYu9BtJ3vm0cbPt5KNZQ@mail.gmail.com>

On Thu, Sep 27, 2012 at 6:32 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> 1. An end-customer calls, the call is handled by mod_ivr.
> 2. Customer is presented with a menu, she makes selection and also enters
> some identification. Her call is parked on a pre-decided extension and an
> entry is made in a database for this call. This entry will hold the
> selections made by the caller and the extension where call is parked.

<SNIP>

> Please guide me on how can do points 1 & 2.

1) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr#General_Concept
2) http://wiki.freeswitch.org/wiki/Mod_db#mod_db ?


Gabe


From teihrib at gmail.com  Thu Sep 27 17:32:56 2012
From: teihrib at gmail.com (=?KOI8-R?Q?=E1nton_=F4eihrib?=)
Date: Thu, 27 Sep 2012 19:32:56 +0600
Subject: [Freeswitch-users] Freeswitch as RTP <=> SRTP translator
Message-ID: <CAMzazSDHA6dB+i95+3iExR7vWtOScts6wh5wfOgGmXr8EuOQMg@mail.gmail.com>

Hi all!
In my case all sip clients uses SRTP, but VoIP provider accepts only RTP
packets. I'm thinking about
using freeswitch between them, like so:
client <=SRTP=> freeswitch <=RTP=> VoIP provider.
Is it possible? If so please give me some direction how to implement this
setup.
Thanks a lot in advance.
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From avi at avimarcus.net  Thu Sep 27 18:43:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 27 Sep 2012 16:43:49 +0200
Subject: [Freeswitch-users] Freeswitch as RTP <=> SRTP translator
In-Reply-To: <CAMzazSDHA6dB+i95+3iExR7vWtOScts6wh5wfOgGmXr8EuOQMg@mail.gmail.com>
References: <CAMzazSDHA6dB+i95+3iExR7vWtOScts6wh5wfOgGmXr8EuOQMg@mail.gmail.com>
Message-ID: <CALNZuwmUUxjQBWDOfMK0-+U72r5Hu3SSoOudtjngoE060x1=DQ@mail.gmail.com>

Surely. Each leg has codecs, secure, etc, set independently so just proceed
as normal.
Set up srtp on the client leg and a normal bridge on the B leg.

Sorry, I can't really give you any more details than that.

-Avi
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From asaad2 at gmail.com  Thu Sep 27 18:55:43 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 27 Sep 2012 10:55:43 -0400
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
Message-ID: <CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>

I had the same issue. There are hackers continuously scanning public ip's
for known ports then trying to register devices using the default
extensions and passwords "1234". After noticing this in my logs I just
changed the default external sip port from 5080 to something else.

Security through obscurity if you will.
P.S. I was also using fail2ban
On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk> wrote:

> Hi There,
>  welcome to our world; hope it didn't cost too much.
> Frontier were pro-active, which is very good. Don't forget to thank them.
> I'd guess that this particular bunch are coming from IP addresses provided
> in the West bank and/or Gaza; that's from where my "visitors" appeared to
> originate.
>
> 1st rule of fight club: Firewalls are no use for a server that is going to
> listen for requests from the Internet and allow authenticated calls to be
> placed from any IP address.
>
> You MUST have reasonable passwords, plus fail2ban is easy to set up and
> works just fine [unless you're using Windoz, in which case God hates you**].
>
> For more refined control (if you know where your external contacts are
> coming from) ...
>
> Consider setting up ACLs (nailing down the IP address ranges from which
> you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition
> there is one place to add in your external correspondents.
>
> Also, consider using cidr= parameters in your directory folder for each of
> your users (if they will only attempt to register or place calls from given
> address ranges).
> Then enable ACLs for incalls in your sip profile(s).
>
> This is all covered on wiki.freeswitch.org -- search for ACLs and take it
> from there.
>
> BTW, you WILL be confused by setting explicit ACLs on registration --
> leave that one commented out until you know what it actually does, as it's
> probably not what you expect. Several strong cups of coffee and protracted
> meditation may help.
>
> Main message:
> -- Immediately - fix the passwords so they're not easy to guess [as the
> bad guys *will* try again and again until they get it right].
> -- set up fail2ban (which has its own page on the wiki) exactly as
> proposed. <======= MOST IMPORTANT
> -- lose the belief that firewalls are going to help protect an
> Internet-listening server as, logically, they can't
> Finally, be amazed at the occasional "block" reports in the fail2ban
> logfile, and wonder how you got away with it for so long.
>
> all the best,
>   Lawrence
> ** There was apparently a talk on how Windows users could get something
> close to a fail2ban-style setup (IIRC, it was on the weekly conf call a
> while back)
>
> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
> > I really think that people give way too much importance to firewalls,
> > specially stateless ones, blocking ports isn't going to do much for you
> > unless you are trying to hide vulnerable services behind it.
> >
> > They used the extension 1000 to make the calls so I would say: activate
> > log-auth-failures on your profile, setup a fail2ban and get stronger
> > passwords.
> >
> > If you want to go further you can use a stateful firewall limiting
> > connections and setup a IDS(recommend snort)
> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
> >
> >>
> >> Hey All,
> >>
> >>
> >> I just got an email from Frontier that there were several attempts to
> >> make international calls.
> >>
> >>
> >> I checked the log file and verified that somehow someone was able to get
> >> access to FS from the internet.
> >>
> >>
> >> here is a sample of the log
> >>
> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/1000 at 50.47.85.167
> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168521352 in context default
> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
> >> [NOTICE] switch_channel.c:941 New Channel
> >> sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168521352 in context default
> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/1000 at 50.47.85.167
> >> [4576bc76-144a-4f6f-8915-871b511c374d]
> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
> >> [4576bc76-144a-4f6f-8915-871b511c374d]
> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168905352 in context default
> >>
> >>
> >> At this point I'm at a loss how this is happening as I have multiple
> >> firewalls in place that limit port access.
> >>
> >> Can someone provide a few pointers on how to better secure FS running on
> >> Linux systems?
> >>
> >>
> >> thanks
> >>
> >>
> >> --
> >> -
> >> -
> >> -    Best Regards,
> >> -
> >> -            Todd Bailey
> >> -
> >> -
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Sep 27 19:38:28 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Thu, 27 Sep 2012 12:38:28 -0300
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
Message-ID: <CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>

This is classic wardialing and is very common. Don't worry, your port
change won't slow down people who really want to get in ;)


On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:

> I had the same issue. There are hackers continuously scanning public ip's
> for known ports then trying to register devices using the default
> extensions and passwords "1234". After noticing this in my logs I just
> changed the default external sip port from 5080 to something else.
>
> Security through obscurity if you will.
> P.S. I was also using fail2ban
> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
> wrote:
>
>> Hi There,
>>  welcome to our world; hope it didn't cost too much.
>> Frontier were pro-active, which is very good. Don't forget to thank them.
>> I'd guess that this particular bunch are coming from IP addresses
>> provided in the West bank and/or Gaza; that's from where my "visitors"
>> appeared to originate.
>>
>> 1st rule of fight club: Firewalls are no use for a server that is going
>> to listen for requests from the Internet and allow authenticated calls to
>> be placed from any IP address.
>>
>> You MUST have reasonable passwords, plus fail2ban is easy to set up and
>> works just fine [unless you're using Windoz, in which case God hates you**].
>>
>> For more refined control (if you know where your external contacts are
>> coming from) ...
>>
>> Consider setting up ACLs (nailing down the IP address ranges from which
>> you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition
>> there is one place to add in your external correspondents.
>>
>> Also, consider using cidr= parameters in your directory folder for each
>> of your users (if they will only attempt to register or place calls from
>> given address ranges).
>> Then enable ACLs for incalls in your sip profile(s).
>>
>> This is all covered on wiki.freeswitch.org -- search for ACLs and take
>> it from there.
>>
>> BTW, you WILL be confused by setting explicit ACLs on registration --
>> leave that one commented out until you know what it actually does, as it's
>> probably not what you expect. Several strong cups of coffee and protracted
>> meditation may help.
>>
>> Main message:
>> -- Immediately - fix the passwords so they're not easy to guess [as the
>> bad guys *will* try again and again until they get it right].
>> -- set up fail2ban (which has its own page on the wiki) exactly as
>> proposed. <======= MOST IMPORTANT
>> -- lose the belief that firewalls are going to help protect an
>> Internet-listening server as, logically, they can't
>> Finally, be amazed at the occasional "block" reports in the fail2ban
>> logfile, and wonder how you got away with it for so long.
>>
>> all the best,
>>   Lawrence
>> ** There was apparently a talk on how Windows users could get something
>> close to a fail2ban-style setup (IIRC, it was on the weekly conf call a
>> while back)
>>
>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>> > I really think that people give way too much importance to firewalls,
>> > specially stateless ones, blocking ports isn't going to do much for you
>> > unless you are trying to hide vulnerable services behind it.
>> >
>> > They used the extension 1000 to make the calls so I would say: activate
>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>> > passwords.
>> >
>> > If you want to go further you can use a stateful firewall limiting
>> > connections and setup a IDS(recommend snort)
>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>> >
>> >>
>> >> Hey All,
>> >>
>> >>
>> >> I just got an email from Frontier that there were several attempts to
>> >> make international calls.
>> >>
>> >>
>> >> I checked the log file and verified that somehow someone was able to
>> get
>> >> access to FS from the internet.
>> >>
>> >>
>> >> here is a sample of the log
>> >>
>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/1000 at 50.47.85.167
>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168521352 in context default
>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
>> >> [NOTICE] switch_channel.c:941 New Channel
>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168521352 in context default
>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/1000 at 50.47.85.167
>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168905352 in context default
>> >>
>> >>
>> >> At this point I'm at a loss how this is happening as I have multiple
>> >> firewalls in place that limit port access.
>> >>
>> >> Can someone provide a few pointers on how to better secure FS running
>> on
>> >> Linux systems?
>> >>
>> >>
>> >> thanks
>> >>
>> >>
>> >> --
>> >> -
>> >> -
>> >> -    Best Regards,
>> >> -
>> >> -            Todd Bailey
>> >> -
>> >> -
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 27 19:52:07 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 27 Sep 2012 11:52:07 -0400
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
Message-ID: <CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>

How will they know what protocol I'm running on that port?
On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:

> This is classic wardialing and is very common. Don't worry, your port
> change won't slow down people who really want to get in ;)
>
>
> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>
>> I had the same issue. There are hackers continuously scanning public ip's
>> for known ports then trying to register devices using the default
>> extensions and passwords "1234". After noticing this in my logs I just
>> changed the default external sip port from 5080 to something else.
>>
>> Security through obscurity if you will.
>> P.S. I was also using fail2ban
>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>> wrote:
>>
>>> Hi There,
>>>  welcome to our world; hope it didn't cost too much.
>>> Frontier were pro-active, which is very good. Don't forget to thank them.
>>> I'd guess that this particular bunch are coming from IP addresses
>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>> appeared to originate.
>>>
>>> 1st rule of fight club: Firewalls are no use for a server that is going
>>> to listen for requests from the Internet and allow authenticated calls to
>>> be placed from any IP address.
>>>
>>> You MUST have reasonable passwords, plus fail2ban is easy to set up and
>>> works just fine [unless you're using Windoz, in which case God hates you**].
>>>
>>> For more refined control (if you know where your external contacts are
>>> coming from) ...
>>>
>>> Consider setting up ACLs (nailing down the IP address ranges from which
>>> you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition
>>> there is one place to add in your external correspondents.
>>>
>>> Also, consider using cidr= parameters in your directory folder for each
>>> of your users (if they will only attempt to register or place calls from
>>> given address ranges).
>>> Then enable ACLs for incalls in your sip profile(s).
>>>
>>> This is all covered on wiki.freeswitch.org -- search for ACLs and take
>>> it from there.
>>>
>>> BTW, you WILL be confused by setting explicit ACLs on registration --
>>> leave that one commented out until you know what it actually does, as it's
>>> probably not what you expect. Several strong cups of coffee and protracted
>>> meditation may help.
>>>
>>> Main message:
>>> -- Immediately - fix the passwords so they're not easy to guess [as the
>>> bad guys *will* try again and again until they get it right].
>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>> proposed. <======= MOST IMPORTANT
>>> -- lose the belief that firewalls are going to help protect an
>>> Internet-listening server as, logically, they can't
>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>> logfile, and wonder how you got away with it for so long.
>>>
>>> all the best,
>>>   Lawrence
>>> ** There was apparently a talk on how Windows users could get something
>>> close to a fail2ban-style setup (IIRC, it was on the weekly conf call a
>>> while back)
>>>
>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>>> > I really think that people give way too much importance to firewalls,
>>> > specially stateless ones, blocking ports isn't going to do much for you
>>> > unless you are trying to hide vulnerable services behind it.
>>> >
>>> > They used the extension 1000 to make the calls so I would say: activate
>>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>>> > passwords.
>>> >
>>> > If you want to go further you can use a stateful firewall limiting
>>> > connections and setup a IDS(recommend snort)
>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>>> >
>>> >>
>>> >> Hey All,
>>> >>
>>> >>
>>> >> I just got an email from Frontier that there were several attempts to
>>> >> make international calls.
>>> >>
>>> >>
>>> >> I checked the log file and verified that somehow someone was able to
>>> get
>>> >> access to FS from the internet.
>>> >>
>>> >>
>>> >> here is a sample of the log
>>> >>
>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>> >> Processing 1000 <1000>->01137168521352 in context default
>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>>> >> sofia/internal/1000 at 50.47.85.167!
>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>> Pre-Answer
>>> >> sofia/internal/1000 at 50.47.85.167!
>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
>>> >> [NOTICE] switch_channel.c:941 New Channel
>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>> >> Processing 1000 <1000>->01137168521352 in context default
>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>>> >> sofia/internal/1000 at 50.47.85.167!
>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>> Pre-Answer
>>> >> sofia/internal/1000 at 50.47.85.167!
>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>> >> Processing 1000 <1000>->01137168905352 in context default
>>> >>
>>> >>
>>> >> At this point I'm at a loss how this is happening as I have multiple
>>> >> firewalls in place that limit port access.
>>> >>
>>> >> Can someone provide a few pointers on how to better secure FS running
>>> on
>>> >> Linux systems?
>>> >>
>>> >>
>>> >> thanks
>>> >>
>>> >>
>>> >> --
>>> >> -
>>> >> -
>>> >> -    Best Regards,
>>> >> -
>>> >> -            Todd Bailey
>>> >> -
>>> >> -
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Sep 27 20:25:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Sep 2012 11:25:39 -0500
Subject: [Freeswitch-users] Sometimes missing a few seconds of audio
 when CN is offered in INVITE (and solution)
In-Reply-To: <1FFF97C269757C458224B7C895F35F15160DC9@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15160DC9@cantor.std.visionutv.se>
Message-ID: <CAKbxfG8B-guiPOA3Ry0qnK++KWO_4C7WRNCAjz0z5X1N9PsD0g@mail.gmail.com>

awesome,

I call that kind of fix a google(tm) commit.

Its the equiv of pressing the [ I'm feeling lucky ] button ;)


On Thu, Sep 27, 2012 at 2:03 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Tony, your change fixed the problem!****
>
> ** **
>
> Thank you so much :)****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
>
> *Skickat:* den 26 september 2012 21:28
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] Sometimes missing a few seconds of audio
> when CN is offered in INVITE (and solution)****
>
> ** **
>
> Looking at the code, I can see a solution assuming your theory is correct.
> ****
>
> I pushed a patch to HEAD if you care to test it.****
>
> ** **
>
> On Wed, Sep 26, 2012 at 3:26 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:****
>
> Hello everyone! I experience a strange issue for bridged calls from Lync
> -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I?m not
> sure if this a bug or not so I ask the question here first.****
>
>  ****
>
> The problem is that the audio back from the Asterisk server is not bridged
> back to the originating client for the first few seconds (how long is
> different on different calls), instead FS seems to send CN-packets back.
> I?ve looked into a wireshark dump, and I see the audio is coming from the
> Asterisk server (using the correct IP and port), but FS doesn?t write the
> same packet to the other call leg (back to Lync), instead it sends a CN
> packet every second or so. My guess is that the timestamp is being handled
> wrong somehow, and FS believes that the real packet is too old (since it
> has already sent a CN packet with a higher timestamp?), and should not be
> written to the other leg, then after a while it sends a new CN packet and
> so on.****
>
>  ****
>
> The solution is to set  ?suppress_cng=true? before bridging the call to
> the Asterisk server, when this is done, the audio is always bridged
> correctly, and nothing is missing.****
>
>  ****
>
> I have pcaps of both working calls (with the variable set) and nonworking
> calls, so if you believe it might be something that FS should handle
> differently I?ll submit this to Jira.****
>
>  ****
>
> /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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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****
>
>  !DSPAM:5063553532761904613329! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Thu Sep 27 20:25:40 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Sep 2012 09:25:40 -0700
Subject: [Freeswitch-users] Empty wiki and my head
In-Reply-To: <CAF=swh_LmBERMqTx4goR4aoGrKfk4ZzMTAh-CoYmGWS90N7Uog@mail.gmail.com>
References: <CAF=swh_LmBERMqTx4goR4aoGrKfk4ZzMTAh-CoYmGWS90N7Uog@mail.gmail.com>
Message-ID: <CAKzWOxUPqjbns7Bo1Zb-2kD_Bb9hK6x=15s0V_As00zZOMO_Ag@mail.gmail.com>

No documentation currently exists, therefore we need community assistance.
I don't believe mod_html5 is complete, so don't sweat that one. I've never
heard of mod_sonar. I know the Moc wrote mod_abstraction and that he talked
about it on a conference call quite a while back but I have no further
information. You might want to track him down (IRC: Moc) and see if he can
give you more info. I do know that mod_xml_scgi is like mod_xml_curl but
using SCGI instead. Again, I've never had a chance to use it, so if anyone
has please step up and help with the documentation.

Feel free to experiment, look at the source code, etc. and put some basic
information on the wiki. I'm sure the dev team would be happy to answer
specific questions. As a show of good faith it might be nice to create the
necessary wiki pages as stubs and then add what you know. Then let the
collective community answer specific questions and fill in the blanks.

-MC



On Tue, Sep 25, 2012 at 7:15 PM, Valery Kalinin <valery.kalinin at gmail.com>wrote:

> I cannot find any documentation for these modules:
> mod_abstraction
> mod_sonar
> mod_html5
> mod_xml_scgi
> What they do and how to use them?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sertys at gmail.com  Thu Sep 27 18:35:31 2012
From: sertys at gmail.com (Daniel Ivanov)
Date: Thu, 27 Sep 2012 17:35:31 +0300
Subject: [Freeswitch-users] Freeswitch as RTP <=> SRTP translator
In-Reply-To: <CAMzazSDHA6dB+i95+3iExR7vWtOScts6wh5wfOgGmXr8EuOQMg@mail.gmail.com>
References: <CAMzazSDHA6dB+i95+3iExR7vWtOScts6wh5wfOgGmXr8EuOQMg@mail.gmail.com>
Message-ID: <CAMeZp609Gz3+hUC2wBBrY7+ebQyUNLBUbK2XGBz+-BLZwCZmHg@mail.gmail.com>

It is possible, just set proxy_media to false and have fs process the
streams and you get independent legs. Your a-leg can talk SRTP, but it's
not necessary that the b-leg talks it. SRTP is user-to-system protection
scheme, not user-to-user.

On Thu, Sep 27, 2012 at 4:32 PM, ?nton ?eihrib <teihrib at gmail.com> wrote:

> Hi all!
> In my case all sip clients uses SRTP, but VoIP provider accepts only RTP
> packets. I'm thinking about
> using freeswitch between them, like so:
> client <=SRTP=> freeswitch <=RTP=> VoIP provider.
> Is it possible? If so please give me some direction how to implement this
> setup.
> Thanks a lot 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 bigx333 at gmail.com  Thu Sep 27 20:11:04 2012
From: bigx333 at gmail.com (Nelson Camargo)
Date: Thu, 27 Sep 2012 18:11:04 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
Message-ID: <F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>

Ever heard about nmap? lol
On 27 Sep 2012, at 5:52 PM, BookBag wrote:

> How will they know what protocol I'm running on that port?
> 
> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
> This is classic wardialing and is very common. Don't worry, your port change won't slow down people who really want to get in ;)
> 
> 
> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
> I had the same issue. There are hackers continuously scanning public ip's for known ports then trying to register devices using the default extensions and passwords "1234". After noticing this in my logs I just changed the default external sip port from 5080 to something else. 
> 
> Security through obscurity if you will. 
> P.S. I was also using fail2ban
> 
> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk> wrote:
> Hi There,
>  welcome to our world; hope it didn't cost too much.
> Frontier were pro-active, which is very good. Don't forget to thank them.
> I'd guess that this particular bunch are coming from IP addresses provided in the West bank and/or Gaza; that's from where my "visitors" appeared to originate.
> 
> 1st rule of fight club: Firewalls are no use for a server that is going to listen for requests from the Internet and allow authenticated calls to be placed from any IP address.
> 
> You MUST have reasonable passwords, plus fail2ban is easy to set up and works just fine [unless you're using Windoz, in which case God hates you**].
> 
> For more refined control (if you know where your external contacts are coming from) ...
> 
> Consider setting up ACLs (nailing down the IP address ranges from which you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition there is one place to add in your external correspondents.
> 
> Also, consider using cidr= parameters in your directory folder for each of your users (if they will only attempt to register or place calls from given address ranges).
> Then enable ACLs for incalls in your sip profile(s).
> 
> This is all covered on wiki.freeswitch.org -- search for ACLs and take it from there.
> 
> BTW, you WILL be confused by setting explicit ACLs on registration -- leave that one commented out until you know what it actually does, as it's probably not what you expect. Several strong cups of coffee and protracted meditation may help.
> 
> Main message:
> -- Immediately - fix the passwords so they're not easy to guess [as the bad guys *will* try again and again until they get it right].
> -- set up fail2ban (which has its own page on the wiki) exactly as proposed. <======= MOST IMPORTANT
> -- lose the belief that firewalls are going to help protect an Internet-listening server as, logically, they can't
> Finally, be amazed at the occasional "block" reports in the fail2ban logfile, and wonder how you got away with it for so long.
> 
> all the best,
>   Lawrence
> ** There was apparently a talk on how Windows users could get something close to a fail2ban-style setup (IIRC, it was on the weekly conf call a while back)
> 
> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
> > I really think that people give way too much importance to firewalls,
> > specially stateless ones, blocking ports isn't going to do much for you
> > unless you are trying to hide vulnerable services behind it.
> >
> > They used the extension 1000 to make the calls so I would say: activate
> > log-auth-failures on your profile, setup a fail2ban and get stronger
> > passwords.
> >
> > If you want to go further you can use a stateful firewall limiting
> > connections and setup a IDS(recommend snort)
> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
> >
> >>
> >> Hey All,
> >>
> >>
> >> I just got an email from Frontier that there were several attempts to
> >> make international calls.
> >>
> >>
> >> I checked the log file and verified that somehow someone was able to get
> >> access to FS from the internet.
> >>
> >>
> >> here is a sample of the log
> >>
> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/1000 at 50.47.85.167
> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168521352 in context default
> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
> >> [NOTICE] switch_channel.c:941 New Channel
> >> sofia/internal/1000 at 50.47.85.167 [af778857-0188-4ed2-a82a-94ae749a02cb]
> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168521352 in context default
> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
> >> sofia/internal/1000 at 50.47.85.167!
> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
> >> Channel sofia/internal/1000 at 50.47.85.167
> >> [4576bc76-144a-4f6f-8915-871b511c374d]
> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
> >> [4576bc76-144a-4f6f-8915-871b511c374d]
> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
> >> Processing 1000 <1000>->01137168905352 in context default
> >>
> >>
> >> At this point I'm at a loss how this is happening as I have multiple
> >> firewalls in place that limit port access.
> >>
> >> Can someone provide a few pointers on how to better secure FS running on
> >> Linux systems?
> >>
> >>
> >> thanks
> >>
> >>
> >> --
> >> -
> >> -
> >> -    Best Regards,
> >> -
> >> -            Todd Bailey
> >> -
> >> -
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From darcy at Vex.Net  Thu Sep 27 20:18:38 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Thu, 27 Sep 2012 12:18:38 -0400
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
Message-ID: <20120927121838.7ea038d8908e8afd6a683533@Vex.Net>

On Thu, 27 Sep 2012 11:52:07 -0400
BookBag <asaad2 at gmail.com> wrote:
> How will they know what protocol I'm running on that port?

They don't have to know.  They just try every protocol on every port.
They are running the attack on some hacked WinBlows box and using your
bandwidth so there is no cost to them.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From msc at freeswitch.org  Thu Sep 27 20:33:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Sep 2012 09:33:27 -0700
Subject: [Freeswitch-users] get events data on freeswitch console
In-Reply-To: <CAN+NqUZnVvm18x7K7rZ+7mt6qphmVuwUetXnPhdzT_tGU84oeA@mail.gmail.com>
References: <CAN+NqUZnVvm18x7K7rZ+7mt6qphmVuwUetXnPhdzT_tGU84oeA@mail.gmail.com>
Message-ID: <CAKzWOxWJJkzLuJ_WHyHFAw1n04vAnRhNsEyYU2mycOO8yzTSnQ@mail.gmail.com>

In this case "params" doesn't seem to have anything. Where does params get
populated in your script?
-MC

On Wed, Sep 26, 2012 at 9:36 AM, Shahzad Bhatti <shahzad.bhatti at g-r-v.com>wrote:

> hi,
>
> i want to get the events data without ESL in lua and with the following:
>
> freeswitch.consoleLog("info",params:serialize())
>
>
> but unable to any output but* ERROR*
>
> 2012-09-27 01:21:47.024894 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/ESLInLua.lua:5: attempt to index global 'params' (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/ESLInLua.lua:5: in main chunk
>
>
> reply me how i can get that.
>
> Regards
>
> Shahzad Bhatti
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 27 20:34:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Sep 2012 09:34:41 -0700
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <D9B39FD4-78E7-42DE-917D-3CAC3E25AF90@kavun.ch>
References: <0AC72351-CABE-41E7-A833-D987748373DB@kavun.ch>
	<CADrcvV9XaUC+BG4weEKss1yTHWqGYesA0Go32EP4M66cLtRLAg@mail.gmail.com>
	<D9B39FD4-78E7-42DE-917D-3CAC3E25AF90@kavun.ch>
Message-ID: <CAKzWOxV9oYZgg9nMJ+-S7uS=whdUVJV29DFLN6nmmo0n+o68mQ@mail.gmail.com>

Does it happen whether you use RFC2833 or inband DTMFs? Just curious.
-MC

On Wed, Sep 26, 2012 at 3:44 PM, Emrah <lists at kavun.ch> wrote:

> Yes I did.
> BTW, the example in the Wiki contradicts the inline documentation in
> switch.xml.
>     <!--
>          The min-dtmf-duration specifies the minimum DTMF duration to use
> on
>          outgoing events. Events shorter than this will be increased in
> duration
>          to match min_dtmf_duration. You cannot configure a dtmf duration
> on a
>          profile that is less than this setting. You may increase this
> value,
>          but cannot set it lower than 400. This value cannot exceed
>          max-dtmf-duration. -->
> The Wiki shows an example with the value at 100.
>
> I tried increasing and decreasing it to no avail, it does not seem to
> interfere with anything I can measure with my ear. :P
> On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com>
> wrote:
>
> > You tried this:
> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
> >
> > On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
> > Hi guys,
> >
> > I am comparing this with an Asterisk and FreeSWITCH installation, using
> the same route, same codecs, same carrier, same phones and same servers? :P
> > I experience a delay when pressing DTMFs on the line that uses
> FreeSWITCH. I am estimating the delay to be around 500 ms.
> >
> > What are the settings I can fine tune to avoid this?
> >
> > All the 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
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep 27 20:35:42 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 27 Sep 2012 18:35:42 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
Message-ID: <CALNZuwm4vO2mTCQuTQEd+FWkQiGyvj=xv3zzP=-uM91R+W5p7g@mail.gmail.com>

nmap, port scanning.
Nearly every port responds in some way, if it's not ignoring that IP.

-Avi


On Thu, Sep 27, 2012 at 5:52 PM, BookBag <asaad2 at gmail.com> wrote:

> How will they know what protocol I'm running on that port?
> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>
>> This is classic wardialing and is very common. Don't worry, your port
>> change won't slow down people who really want to get in ;)
>>
>>
>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>
>>> I had the same issue. There are hackers continuously scanning public
>>> ip's for known ports then trying to register devices using the default
>>> extensions and passwords "1234". After noticing this in my logs I just
>>> changed the default external sip port from 5080 to something else.
>>>
>>> Security through obscurity if you will.
>>> P.S. I was also using fail2ban
>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>> wrote:
>>>
>>>> Hi There,
>>>>  welcome to our world; hope it didn't cost too much.
>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>> them.
>>>> I'd guess that this particular bunch are coming from IP addresses
>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>> appeared to originate.
>>>>
>>>> 1st rule of fight club: Firewalls are no use for a server that is going
>>>> to listen for requests from the Internet and allow authenticated calls to
>>>> be placed from any IP address.
>>>>
>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up and
>>>> works just fine [unless you're using Windoz, in which case God hates you**].
>>>>
>>>> For more refined control (if you know where your external contacts are
>>>> coming from) ...
>>>>
>>>> Consider setting up ACLs (nailing down the IP address ranges from which
>>>> you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition
>>>> there is one place to add in your external correspondents.
>>>>
>>>> Also, consider using cidr= parameters in your directory folder for each
>>>> of your users (if they will only attempt to register or place calls from
>>>> given address ranges).
>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>
>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and take
>>>> it from there.
>>>>
>>>> BTW, you WILL be confused by setting explicit ACLs on registration --
>>>> leave that one commented out until you know what it actually does, as it's
>>>> probably not what you expect. Several strong cups of coffee and protracted
>>>> meditation may help.
>>>>
>>>> Main message:
>>>> -- Immediately - fix the passwords so they're not easy to guess [as the
>>>> bad guys *will* try again and again until they get it right].
>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>> proposed. <======= MOST IMPORTANT
>>>> -- lose the belief that firewalls are going to help protect an
>>>> Internet-listening server as, logically, they can't
>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>> logfile, and wonder how you got away with it for so long.
>>>>
>>>> all the best,
>>>>   Lawrence
>>>> ** There was apparently a talk on how Windows users could get something
>>>> close to a fail2ban-style setup (IIRC, it was on the weekly conf call a
>>>> while back)
>>>>
>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>>>> > I really think that people give way too much importance to firewalls,
>>>> > specially stateless ones, blocking ports isn't going to do much for
>>>> you
>>>> > unless you are trying to hide vulnerable services behind it.
>>>> >
>>>> > They used the extension 1000 to make the calls so I would say:
>>>> activate
>>>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>>>> > passwords.
>>>> >
>>>> > If you want to go further you can use a stateful firewall limiting
>>>> > connections and setup a IDS(recommend snort)
>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>>>> >
>>>> >>
>>>> >> Hey All,
>>>> >>
>>>> >>
>>>> >> I just got an email from Frontier that there were several attempts to
>>>> >> make international calls.
>>>> >>
>>>> >>
>>>> >> I checked the log file and verified that somehow someone was able to
>>>> get
>>>> >> access to FS from the internet.
>>>> >>
>>>> >>
>>>> >> here is a sample of the log
>>>> >>
>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>> Ring-Ready
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>> Pre-Answer
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>> switch_ivr_originate.c:3303
>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>> 16:30:29.916821
>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>> Ring-Ready
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>> Pre-Answer
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>> switch_ivr_originate.c:3303
>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>> >>
>>>> >>
>>>> >> At this point I'm at a loss how this is happening as I have multiple
>>>> >> firewalls in place that limit port access.
>>>> >>
>>>> >> Can someone provide a few pointers on how to better secure FS
>>>> running on
>>>> >> Linux systems?
>>>> >>
>>>> >>
>>>> >> thanks
>>>> >>
>>>> >>
>>>> >> --
>>>> >> -
>>>> >> -
>>>> >> -    Best Regards,
>>>> >> -
>>>> >> -            Todd Bailey
>>>> >> -
>>>> >> -
>>>> >>
>>>> >>
>>>> >>
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> 
>>>> >> 
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://wiki.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > 
>>>> > 
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://wiki.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Thu Sep 27 20:36:40 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 27 Sep 2012 18:36:40 +0200
Subject: [Freeswitch-users] Freeswitch as RTP <=> SRTP translator
In-Reply-To: <CAMeZp609Gz3+hUC2wBBrY7+ebQyUNLBUbK2XGBz+-BLZwCZmHg@mail.gmail.com>
References: <CAMzazSDHA6dB+i95+3iExR7vWtOScts6wh5wfOgGmXr8EuOQMg@mail.gmail.com>
	<CAMeZp609Gz3+hUC2wBBrY7+ebQyUNLBUbK2XGBz+-BLZwCZmHg@mail.gmail.com>
Message-ID: <CALNZuwnb9UzJwOXC8Bjrqm5wFk0oT0HRN-d8waobU5VvWpz_=g@mail.gmail.com>

... proxy_media is false by default.
-Avi



On Thu, Sep 27, 2012 at 4:35 PM, Daniel Ivanov <sertys at gmail.com> wrote:

> It is possible, just set proxy_media to false and have fs process the
> streams and you get independent legs. Your a-leg can talk SRTP, but it's
> not necessary that the b-leg talks it. SRTP is user-to-system protection
> scheme, not user-to-user.
>
> On Thu, Sep 27, 2012 at 4:32 PM, ?nton ?eihrib <teihrib at gmail.com> wrote:
>
>> Hi all!
>> In my case all sip clients uses SRTP, but VoIP provider accepts only RTP
>> packets. I'm thinking about
>> using freeswitch between them, like so:
>> client <=SRTP=> freeswitch <=RTP=> VoIP provider.
>> Is it possible? If so please give me some direction how to implement this
>> setup.
>> Thanks a lot 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 lists at kavun.ch  Thu Sep 27 20:46:38 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 27 Sep 2012 12:46:38 -0400
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <CAKzWOxV9oYZgg9nMJ+-S7uS=whdUVJV29DFLN6nmmo0n+o68mQ@mail.gmail.com>
References: <0AC72351-CABE-41E7-A833-D987748373DB@kavun.ch>
	<CADrcvV9XaUC+BG4weEKss1yTHWqGYesA0Go32EP4M66cLtRLAg@mail.gmail.com>
	<D9B39FD4-78E7-42DE-917D-3CAC3E25AF90@kavun.ch>
	<CAKzWOxV9oYZgg9nMJ+-S7uS=whdUVJV29DFLN6nmmo0n+o68mQ@mail.gmail.com>
Message-ID: <B293CD40-30DA-4FE7-BBB0-37BD0EB194BA@kavun.ch>

Never tried with inband DTMFs. Will check.

Thanks!
On Sep 27, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:

> Does it happen whether you use RFC2833 or inband DTMFs? Just curious.
> -MC
> 
> On Wed, Sep 26, 2012 at 3:44 PM, Emrah <lists at kavun.ch> wrote:
> Yes I did.
> BTW, the example in the Wiki contradicts the inline documentation in switch.xml.
>     <!--
>          The min-dtmf-duration specifies the minimum DTMF duration to use on
>          outgoing events. Events shorter than this will be increased in duration
>          to match min_dtmf_duration. You cannot configure a dtmf duration on a
>          profile that is less than this setting. You may increase this value,
>          but cannot set it lower than 400. This value cannot exceed
>          max-dtmf-duration. -->
> The Wiki shows an example with the value at 100.
> 
> I tried increasing and decreasing it to no avail, it does not seem to interfere with anything I can measure with my ear. :P
> On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com> wrote:
> 
> > You tried this:
> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
> >
> > On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
> > Hi guys,
> >
> > I am comparing this with an Asterisk and FreeSWITCH installation, using the same route, same codecs, same carrier, same phones and same servers? :P
> > I experience a delay when pressing DTMFs on the line that uses FreeSWITCH. I am estimating the delay to be around 500 ms.
> >
> > What are the settings I can fine tune to avoid this?
> >
> > All the 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
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Thu Sep 27 20:52:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Sep 2012 09:52:01 -0700
Subject: [Freeswitch-users] Help needed in a Contact Center solution
In-Reply-To: <CAMwZ-tJnaHnwq1P+Z91O+_vz54xmPjdYu9BtJ3vm0cbPt5KNZQ@mail.gmail.com>
References: <011601cd9cac$2919d840$7b4d88c0$@co.in>
	<CAMwZ-tJnaHnwq1P+Z91O+_vz54xmPjdYu9BtJ3vm0cbPt5KNZQ@mail.gmail.com>
Message-ID: <CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>

Don't forget about our two books:
http://link.packtpub.com/nuIOlX
https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book

They discuss items 1 and 2 in pretty good detail, especially chapter 6 of
the "bridge" book (it has a bridge on the cover)
-MC

On Thu, Sep 27, 2012 at 5:53 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, Sep 27, 2012 at 6:32 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> > 1. An end-customer calls, the call is handled by mod_ivr.
> > 2. Customer is presented with a menu, she makes selection and also enters
> > some identification. Her call is parked on a pre-decided extension and an
> > entry is made in a database for this call. This entry will hold the
> > selections made by the caller and the extension where call is parked.
>
> <SNIP>
>
> > Please guide me on how can do points 1 & 2.
>
> 1)
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr#General_Concept
> 2) http://wiki.freeswitch.org/wiki/Mod_db#mod_db ?
>
>
> 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
>



-- 
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 Sep 27 20:57:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Sep 2012 09:57:58 -0700
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
Message-ID: <CAKzWOxXSyi7z_zkgvWh2WaH3xxUDsVV_brnoVkkHxjkw3qi5tg@mail.gmail.com>

FYI, don't forget this:
$src/scripts/perl/randomize_passwords.pl

It's useful if you're stuck using static XML files and you want a quick and
easy way to mix things up. Also, if you are using static XML you may find
this handy:
$src/scripts/perl/add_user

Simple way to add users with longer vm passwords and scrambled SIP auth
passwords.

-MC

On Thu, Sep 27, 2012 at 7:55 AM, BookBag <asaad2 at gmail.com> wrote:

> I had the same issue. There are hackers continuously scanning public ip's
> for known ports then trying to register devices using the default
> extensions and passwords "1234". After noticing this in my logs I just
> changed the default external sip port from 5080 to something else.
>
> Security through obscurity if you will.
> P.S. I was also using fail2ban
> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
> wrote:
>
>> Hi There,
>>  welcome to our world; hope it didn't cost too much.
>> Frontier were pro-active, which is very good. Don't forget to thank them.
>> I'd guess that this particular bunch are coming from IP addresses
>> provided in the West bank and/or Gaza; that's from where my "visitors"
>> appeared to originate.
>>
>> 1st rule of fight club: Firewalls are no use for a server that is going
>> to listen for requests from the Internet and allow authenticated calls to
>> be placed from any IP address.
>>
>> You MUST have reasonable passwords, plus fail2ban is easy to set up and
>> works just fine [unless you're using Windoz, in which case God hates you**].
>>
>> For more refined control (if you know where your external contacts are
>> coming from) ...
>>
>> Consider setting up ACLs (nailing down the IP address ranges from which
>> you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition
>> there is one place to add in your external correspondents.
>>
>> Also, consider using cidr= parameters in your directory folder for each
>> of your users (if they will only attempt to register or place calls from
>> given address ranges).
>> Then enable ACLs for incalls in your sip profile(s).
>>
>> This is all covered on wiki.freeswitch.org -- search for ACLs and take
>> it from there.
>>
>> BTW, you WILL be confused by setting explicit ACLs on registration --
>> leave that one commented out until you know what it actually does, as it's
>> probably not what you expect. Several strong cups of coffee and protracted
>> meditation may help.
>>
>> Main message:
>> -- Immediately - fix the passwords so they're not easy to guess [as the
>> bad guys *will* try again and again until they get it right].
>> -- set up fail2ban (which has its own page on the wiki) exactly as
>> proposed. <======= MOST IMPORTANT
>> -- lose the belief that firewalls are going to help protect an
>> Internet-listening server as, logically, they can't
>> Finally, be amazed at the occasional "block" reports in the fail2ban
>> logfile, and wonder how you got away with it for so long.
>>
>> all the best,
>>   Lawrence
>> ** There was apparently a talk on how Windows users could get something
>> close to a fail2ban-style setup (IIRC, it was on the weekly conf call a
>> while back)
>>
>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>> > I really think that people give way too much importance to firewalls,
>> > specially stateless ones, blocking ports isn't going to do much for you
>> > unless you are trying to hide vulnerable services behind it.
>> >
>> > They used the extension 1000 to make the calls so I would say: activate
>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>> > passwords.
>> >
>> > If you want to go further you can use a stateful firewall limiting
>> > connections and setup a IDS(recommend snort)
>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>> >
>> >>
>> >> Hey All,
>> >>
>> >>
>> >> I just got an email from Frontier that there were several attempts to
>> >> make international calls.
>> >>
>> >>
>> >> I checked the log file and verified that somehow someone was able to
>> get
>> >> access to FS from the internet.
>> >>
>> >>
>> >> here is a sample of the log
>> >>
>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/1000 at 50.47.85.167
>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168521352 in context default
>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23 16:30:29.916821
>> >> [NOTICE] switch_channel.c:941 New Channel
>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168521352 in context default
>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572 Ring-Ready
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176 Pre-Answer
>> >> sofia/internal/1000 at 50.47.85.167!
>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] switch_ivr_originate.c:3303
>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>> >> Channel sofia/internal/1000 at 50.47.85.167
>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>> >> Processing 1000 <1000>->01137168905352 in context default
>> >>
>> >>
>> >> At this point I'm at a loss how this is happening as I have multiple
>> >> firewalls in place that limit port access.
>> >>
>> >> Can someone provide a few pointers on how to better secure FS running
>> on
>> >> Linux systems?
>> >>
>> >>
>> >> thanks
>> >>
>> >>
>> >> --
>> >> -
>> >> -
>> >> -    Best Regards,
>> >> -
>> >> -            Todd Bailey
>> >> -
>> >> -
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gabe at gundy.org  Thu Sep 27 22:29:30 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 27 Sep 2012 12:29:30 -0600
Subject: [Freeswitch-users] Help needed in a Contact Center solution
In-Reply-To: <CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>
References: <011601cd9cac$2919d840$7b4d88c0$@co.in>
	<CAMwZ-tJnaHnwq1P+Z91O+_vz54xmPjdYu9BtJ3vm0cbPt5KNZQ@mail.gmail.com>
	<CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>
Message-ID: <CAMwZ-tKGYdbEBfgG-OR9321Hfm-Z7N=KrmvBPfXh3Dcf7ZVT0A@mail.gmail.com>

On Thu, Sep 27, 2012 at 10:52 AM, Michael Collins <msc at freeswitch.org> wrote:
> Don't forget about our two books:
> http://link.packtpub.com/nuIOlX
> https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book
>
> They discuss items 1 and 2 in pretty good detail, especially chapter 6 of
> the "bridge" book (it has a bridge on the cover)

Oh wait... I'm I the *last* person in the world to realize that the
'bridge' is on the cover of the book because of the 'bridge' app?  Oh
man!


Gabe


From lists at kavun.ch  Thu Sep 27 22:49:52 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 27 Sep 2012 14:49:52 -0400
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <B293CD40-30DA-4FE7-BBB0-37BD0EB194BA@kavun.ch>
References: <0AC72351-CABE-41E7-A833-D987748373DB@kavun.ch>
	<CADrcvV9XaUC+BG4weEKss1yTHWqGYesA0Go32EP4M66cLtRLAg@mail.gmail.com>
	<D9B39FD4-78E7-42DE-917D-3CAC3E25AF90@kavun.ch>
	<CAKzWOxV9oYZgg9nMJ+-S7uS=whdUVJV29DFLN6nmmo0n+o68mQ@mail.gmail.com>
	<B293CD40-30DA-4FE7-BBB0-37BD0EB194BA@kavun.ch>
Message-ID: <859D5035-A9F7-4B32-A83E-FD5588AAEFF2@kavun.ch>

MC, the issue does not happen with inband DTMF and there is no delay!

Any idea on how to debug this further? I can't use inband continuously.

Thanks!
Emrah

On Sep 27, 2012, at 12:46 PM, Emrah <lists at kavun.ch> wrote:

> Never tried with inband DTMFs. Will check.
> 
> Thanks!
> On Sep 27, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> Does it happen whether you use RFC2833 or inband DTMFs? Just curious.
>> -MC
>> 
>> On Wed, Sep 26, 2012 at 3:44 PM, Emrah <lists at kavun.ch> wrote:
>> Yes I did.
>> BTW, the example in the Wiki contradicts the inline documentation in switch.xml.
>>    <!--
>>         The min-dtmf-duration specifies the minimum DTMF duration to use on
>>         outgoing events. Events shorter than this will be increased in duration
>>         to match min_dtmf_duration. You cannot configure a dtmf duration on a
>>         profile that is less than this setting. You may increase this value,
>>         but cannot set it lower than 400. This value cannot exceed
>>         max-dtmf-duration. -->
>> The Wiki shows an example with the value at 100.
>> 
>> I tried increasing and decreasing it to no avail, it does not seem to interfere with anything I can measure with my ear. :P
>> On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com> wrote:
>> 
>>> You tried this:
>>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
>>> 
>>> On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
>>> Hi guys,
>>> 
>>> I am comparing this with an Asterisk and FreeSWITCH installation, using the same route, same codecs, same carrier, same phones and same servers? :P
>>> I experience a delay when pressing DTMFs on the line that uses FreeSWITCH. I am estimating the delay to be around 500 ms.
>>> 
>>> What are the settings I can fine tune to avoid this?
>>> 
>>> All the 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
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 krice at freeswitch.org  Thu Sep 27 23:00:18 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Sep 2012 14:00:18 -0500
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <859D5035-A9F7-4B32-A83E-FD5588AAEFF2@kavun.ch>
Message-ID: <CC8A0C72.54439%krice@freeswitch.org>

There can be a delay of DTMF in and DTMF out if you are sending long DTMFs
using 2833, FreeSWITCH gets the entire DMTF and duration then regenerates
it...

If you don't need to interpret the DTMF you can set a variable to make it
just pass the DTMF through untouched... But this has its own set of caveats
(ie: if whatever is sending you DTMF is broken it just pass broken 2833
DTMF)

See http://wiki.freeswitch.org/wiki/Variable_pass_rfc2833



On 9/27/12 1:49 PM, "Emrah" <lists at kavun.ch> wrote:

> MC, the issue does not happen with inband DTMF and there is no delay!
> 
> Any idea on how to debug this further? I can't use inband continuously.
> 
> Thanks!
> Emrah
> 
> On Sep 27, 2012, at 12:46 PM, Emrah <lists at kavun.ch> wrote:
> 
>> Never tried with inband DTMFs. Will check.
>> 
>> Thanks!
>> On Sep 27, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> Does it happen whether you use RFC2833 or inband DTMFs? Just curious.
>>> -MC
>>> 
>>> On Wed, Sep 26, 2012 at 3:44 PM, Emrah <lists at kavun.ch> wrote:
>>> Yes I did.
>>> BTW, the example in the Wiki contradicts the inline documentation in
>>> switch.xml.
>>>    <!--
>>>         The min-dtmf-duration specifies the minimum DTMF duration to use on
>>>         outgoing events. Events shorter than this will be increased in
>>> duration
>>>         to match min_dtmf_duration. You cannot configure a dtmf duration on
>>> a
>>>         profile that is less than this setting. You may increase this value,
>>>         but cannot set it lower than 400. This value cannot exceed
>>>         max-dtmf-duration. -->
>>> The Wiki shows an example with the value at 100.
>>> 
>>> I tried increasing and decreasing it to no avail, it does not seem to
>>> interfere with anything I can measure with my ear. :P
>>> On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com>
>>> wrote:
>>> 
>>>> You tried this:
>>>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
>>>> 
>>>> On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
>>>> Hi guys,
>>>> 
>>>> I am comparing this with an Asterisk and FreeSWITCH installation, using the
>>>> same route, same codecs, same carrier, same phones and same servers? :P
>>>> I experience a delay when pressing DTMFs on the line that uses FreeSWITCH.
>>>> I am estimating the delay to be around 500 ms.
>>>> 
>>>> What are the settings I can fine tune to avoid this?
>>>> 
>>>> All the 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
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From teihrib at gmail.com  Thu Sep 27 22:15:14 2012
From: teihrib at gmail.com (=?KOI8-R?Q?=E1nton_=F4eihrib?=)
Date: Fri, 28 Sep 2012 00:15:14 +0600
Subject: [Freeswitch-users] Freeswitch as RTP <=> SRTP translator
In-Reply-To: <CAMeZp609Gz3+hUC2wBBrY7+ebQyUNLBUbK2XGBz+-BLZwCZmHg@mail.gmail.com>
References: <CAMzazSDHA6dB+i95+3iExR7vWtOScts6wh5wfOgGmXr8EuOQMg@mail.gmail.com>
	<CAMeZp609Gz3+hUC2wBBrY7+ebQyUNLBUbK2XGBz+-BLZwCZmHg@mail.gmail.com>
Message-ID: <CAMzazSCFHLCMxZ2TK3k82ZWNK=UifLRXzbXi+Jx5TunVF=02Xw@mail.gmail.com>

Ok, nice to hear it. Many thanks to you guys!
Another question (maybe dumb) about this setup.
Is it necessary to freeswitch to handle registrations or FS can forward
them to VoIP provider?
I mean, for example, user A and user B have sip accounts at voip provider
and they can register at voip provider through fs.
Or I have to register them in fs independently?

2012/9/27 Daniel Ivanov <sertys at gmail.com>

> It is possible, just set proxy_media to false and have fs process the
> streams and you get independent legs. Your a-leg can talk SRTP, but it's
> not necessary that the b-leg talks it. SRTP is user-to-system protection
> scheme, not user-to-user.
>
> On Thu, Sep 27, 2012 at 4:32 PM, ?nton ?eihrib <teihrib at gmail.com> wrote:
>
>> Hi all!
>> In my case all sip clients uses SRTP, but VoIP provider accepts only RTP
>> packets. I'm thinking about
>> using freeswitch between them, like so:
>> client <=SRTP=> freeswitch <=RTP=> VoIP provider.
>> Is it possible? If so please give me some direction how to implement this
>> setup.
>> Thanks a lot 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  Thu Sep 27 23:27:48 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 27 Sep 2012 15:27:48 -0400
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
Message-ID: <CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>

when nmap finds a port open, it looks in its database of what protocol is
likely to be running on that port. It doesnt actually test if the standard
protocol is running on that port.





On Thu, Sep 27, 2012 at 12:11 PM, Nelson Camargo <bigx333 at gmail.com> wrote:

> Ever heard about nmap? lol
> On 27 Sep 2012, at 5:52 PM, BookBag wrote:
>
> How will they know what protocol I'm running on that port?
> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>
>> This is classic wardialing and is very common. Don't worry, your port
>> change won't slow down people who really want to get in ;)
>>
>>
>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>
>>> I had the same issue. There are hackers continuously scanning public
>>> ip's for known ports then trying to register devices using the default
>>> extensions and passwords "1234". After noticing this in my logs I just
>>> changed the default external sip port from 5080 to something else.
>>>
>>> Security through obscurity if you will.
>>> P.S. I was also using fail2ban
>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>> wrote:
>>>
>>>> Hi There,
>>>>  welcome to our world; hope it didn't cost too much.
>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>> them.
>>>> I'd guess that this particular bunch are coming from IP addresses
>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>> appeared to originate.
>>>>
>>>> 1st rule of fight club: Firewalls are no use for a server that is going
>>>> to listen for requests from the Internet and allow authenticated calls to
>>>> be placed from any IP address.
>>>>
>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up and
>>>> works just fine [unless you're using Windoz, in which case God hates you**].
>>>>
>>>> For more refined control (if you know where your external contacts are
>>>> coming from) ...
>>>>
>>>> Consider setting up ACLs (nailing down the IP address ranges from which
>>>> you'll accept incalls) in autoload/acl.conf.xml -- the "domains" definition
>>>> there is one place to add in your external correspondents.
>>>>
>>>> Also, consider using cidr= parameters in your directory folder for each
>>>> of your users (if they will only attempt to register or place calls from
>>>> given address ranges).
>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>
>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and take
>>>> it from there.
>>>>
>>>> BTW, you WILL be confused by setting explicit ACLs on registration --
>>>> leave that one commented out until you know what it actually does, as it's
>>>> probably not what you expect. Several strong cups of coffee and protracted
>>>> meditation may help.
>>>>
>>>> Main message:
>>>> -- Immediately - fix the passwords so they're not easy to guess [as the
>>>> bad guys *will* try again and again until they get it right].
>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>> proposed. <======= MOST IMPORTANT
>>>> -- lose the belief that firewalls are going to help protect an
>>>> Internet-listening server as, logically, they can't
>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>> logfile, and wonder how you got away with it for so long.
>>>>
>>>> all the best,
>>>>   Lawrence
>>>> ** There was apparently a talk on how Windows users could get something
>>>> close to a fail2ban-style setup (IIRC, it was on the weekly conf call a
>>>> while back)
>>>>
>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>>>> > I really think that people give way too much importance to firewalls,
>>>> > specially stateless ones, blocking ports isn't going to do much for
>>>> you
>>>> > unless you are trying to hide vulnerable services behind it.
>>>> >
>>>> > They used the extension 1000 to make the calls so I would say:
>>>> activate
>>>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>>>> > passwords.
>>>> >
>>>> > If you want to go further you can use a stateful firewall limiting
>>>> > connections and setup a IDS(recommend snort)
>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>>>> >
>>>> >>
>>>> >> Hey All,
>>>> >>
>>>> >>
>>>> >> I just got an email from Frontier that there were several attempts to
>>>> >> make international calls.
>>>> >>
>>>> >>
>>>> >> I checked the log file and verified that somehow someone was able to
>>>> get
>>>> >> access to FS from the internet.
>>>> >>
>>>> >>
>>>> >> here is a sample of the log
>>>> >>
>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>> Ring-Ready
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>> Pre-Answer
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>> switch_ivr_originate.c:3303
>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>> 16:30:29.916821
>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>> Ring-Ready
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] switch_ivr_originate.c:519
>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>> Pre-Answer
>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>> switch_ivr_originate.c:3303
>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>> >>
>>>> >>
>>>> >> At this point I'm at a loss how this is happening as I have multiple
>>>> >> firewalls in place that limit port access.
>>>> >>
>>>> >> Can someone provide a few pointers on how to better secure FS
>>>> running on
>>>> >> Linux systems?
>>>> >>
>>>> >>
>>>> >> thanks
>>>> >>
>>>> >>
>>>> >> --
>>>> >> -
>>>> >> -
>>>> >> -    Best Regards,
>>>> >> -
>>>> >> -            Todd Bailey
>>>> >> -
>>>> >> -
>>>> >>
>>>> >>
>>>> >>
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> 
>>>> >> 
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://wiki.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > 
>>>> > 
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://wiki.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Sep 27 23:39:03 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Thu, 27 Sep 2012 16:39:03 -0300
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
	<CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
Message-ID: <CAAyX+KHheAzeLUFLvaBu0gY2wvj3Y43hFqZBkYUh2qQfTcoV1w@mail.gmail.com>

nmap is not a wardialer. If I want to know if you're running SIP, I'll just
try and talk SIP to you on every port that's open, and see if I get
something that's not garbage back. It's brute force, nothing more, and can
be done very performantly.

Regards,
Ben Langfeld


On 27 September 2012 16:27, BookBag <asaad2 at gmail.com> wrote:

> when nmap finds a port open, it looks in its database of what protocol is
> likely to be running on that port. It doesnt actually test if the standard
> protocol is running on that port.
>
>
>
>
>
> On Thu, Sep 27, 2012 at 12:11 PM, Nelson Camargo <bigx333 at gmail.com>wrote:
>
>> Ever heard about nmap? lol
>> On 27 Sep 2012, at 5:52 PM, BookBag wrote:
>>
>> How will they know what protocol I'm running on that port?
>> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>
>>> This is classic wardialing and is very common. Don't worry, your port
>>> change won't slow down people who really want to get in ;)
>>>
>>>
>>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>>
>>>> I had the same issue. There are hackers continuously scanning public
>>>> ip's for known ports then trying to register devices using the default
>>>> extensions and passwords "1234". After noticing this in my logs I just
>>>> changed the default external sip port from 5080 to something else.
>>>>
>>>> Security through obscurity if you will.
>>>> P.S. I was also using fail2ban
>>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>>> wrote:
>>>>
>>>>> Hi There,
>>>>>  welcome to our world; hope it didn't cost too much.
>>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>>> them.
>>>>> I'd guess that this particular bunch are coming from IP addresses
>>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>>> appeared to originate.
>>>>>
>>>>> 1st rule of fight club: Firewalls are no use for a server that is
>>>>> going to listen for requests from the Internet and allow authenticated
>>>>> calls to be placed from any IP address.
>>>>>
>>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up
>>>>> and works just fine [unless you're using Windoz, in which case God hates
>>>>> you**].
>>>>>
>>>>> For more refined control (if you know where your external contacts are
>>>>> coming from) ...
>>>>>
>>>>> Consider setting up ACLs (nailing down the IP address ranges from
>>>>> which you'll accept incalls) in autoload/acl.conf.xml -- the "domains"
>>>>> definition there is one place to add in your external correspondents.
>>>>>
>>>>> Also, consider using cidr= parameters in your directory folder for
>>>>> each of your users (if they will only attempt to register or place calls
>>>>> from given address ranges).
>>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>>
>>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and
>>>>> take it from there.
>>>>>
>>>>> BTW, you WILL be confused by setting explicit ACLs on registration --
>>>>> leave that one commented out until you know what it actually does, as it's
>>>>> probably not what you expect. Several strong cups of coffee and protracted
>>>>> meditation may help.
>>>>>
>>>>> Main message:
>>>>> -- Immediately - fix the passwords so they're not easy to guess [as
>>>>> the bad guys *will* try again and again until they get it right].
>>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>>> proposed. <======= MOST IMPORTANT
>>>>> -- lose the belief that firewalls are going to help protect an
>>>>> Internet-listening server as, logically, they can't
>>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>>> logfile, and wonder how you got away with it for so long.
>>>>>
>>>>> all the best,
>>>>>   Lawrence
>>>>> ** There was apparently a talk on how Windows users could get
>>>>> something close to a fail2ban-style setup (IIRC, it was on the weekly conf
>>>>> call a while back)
>>>>>
>>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>>>>> > I really think that people give way too much importance to firewalls,
>>>>> > specially stateless ones, blocking ports isn't going to do much for
>>>>> you
>>>>> > unless you are trying to hide vulnerable services behind it.
>>>>> >
>>>>> > They used the extension 1000 to make the calls so I would say:
>>>>> activate
>>>>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>>>>> > passwords.
>>>>> >
>>>>> > If you want to go further you can use a stateful firewall limiting
>>>>> > connections and setup a IDS(recommend snort)
>>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>>>>> >
>>>>> >>
>>>>> >> Hey All,
>>>>> >>
>>>>> >>
>>>>> >> I just got an email from Frontier that there were several attempts
>>>>> to
>>>>> >> make international calls.
>>>>> >>
>>>>> >>
>>>>> >> I checked the log file and verified that somehow someone was able
>>>>> to get
>>>>> >> access to FS from the internet.
>>>>> >>
>>>>> >>
>>>>> >> here is a sample of the log
>>>>> >>
>>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>> Ring-Ready
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>> switch_ivr_originate.c:519
>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>> Pre-Answer
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>> switch_ivr_originate.c:3303
>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>>> 16:30:29.916821
>>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>> Ring-Ready
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>> switch_ivr_originate.c:519
>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>> Pre-Answer
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>> switch_ivr_originate.c:3303
>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>>> >>
>>>>> >>
>>>>> >> At this point I'm at a loss how this is happening as I have multiple
>>>>> >> firewalls in place that limit port access.
>>>>> >>
>>>>> >> Can someone provide a few pointers on how to better secure FS
>>>>> running on
>>>>> >> Linux systems?
>>>>> >>
>>>>> >>
>>>>> >> thanks
>>>>> >>
>>>>> >>
>>>>> >> --
>>>>> >> -
>>>>> >> -
>>>>> >> -    Best Regards,
>>>>> >> -
>>>>> >> -            Todd Bailey
>>>>> >> -
>>>>> >> -
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> _________________________________________________________________________
>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>> >> consulting at freeswitch.org
>>>>> >> http://www.freeswitchsolutions.com
>>>>> >>
>>>>> >> 
>>>>> >> 
>>>>> >>
>>>>> >> Official FreeSWITCH Sites
>>>>> >> http://www.freeswitch.org
>>>>> >> http://wiki.freeswitch.org
>>>>> >> http://www.cluecon.com
>>>>> >>
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > 
>>>>> > 
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://wiki.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Thu Sep 27 23:53:10 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 27 Sep 2012 21:53:10 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
	<CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
Message-ID: <CALNZuwntPp-h9VJELy++T+8wsWn4mf+3u-HSwuq+S5TwdYZynw@mail.gmail.com>

nmap offers service detection:


# nmap -sV some-domain.com
...
22/tcp   open   ssh         OpenSSH 5.3p1 Debian 3ubuntu7 (protocol 2.0)
80/tcp   open   http        nginx web server 0.8.54
...
5060/tcp open   sip         (SIP end point; Status: 200 OK)
5080/tcp open   sip         (SIP end point; Status: 200 OK)
...
Nmap done: 1 IP address (1 host up) scanned in 90.91 seconds

vs 5 seconds for plain scan. But still, it exists.

-Avi



On Thu, Sep 27, 2012 at 9:27 PM, BookBag <asaad2 at gmail.com> wrote:

> when nmap finds a port open, it looks in its database of what protocol is
> likely to be running on that port. It doesnt actually test if the standard
> protocol is running on that port.
>
>
>
>
>
> On Thu, Sep 27, 2012 at 12:11 PM, Nelson Camargo <bigx333 at gmail.com>wrote:
>
>> Ever heard about nmap? lol
>> On 27 Sep 2012, at 5:52 PM, BookBag wrote:
>>
>> How will they know what protocol I'm running on that port?
>> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>
>>> This is classic wardialing and is very common. Don't worry, your port
>>> change won't slow down people who really want to get in ;)
>>>
>>>
>>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>>
>>>> I had the same issue. There are hackers continuously scanning public
>>>> ip's for known ports then trying to register devices using the default
>>>> extensions and passwords "1234". After noticing this in my logs I just
>>>> changed the default external sip port from 5080 to something else.
>>>>
>>>> Security through obscurity if you will.
>>>> P.S. I was also using fail2ban
>>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>>> wrote:
>>>>
>>>>> Hi There,
>>>>>  welcome to our world; hope it didn't cost too much.
>>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>>> them.
>>>>> I'd guess that this particular bunch are coming from IP addresses
>>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>>> appeared to originate.
>>>>>
>>>>> 1st rule of fight club: Firewalls are no use for a server that is
>>>>> going to listen for requests from the Internet and allow authenticated
>>>>> calls to be placed from any IP address.
>>>>>
>>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up
>>>>> and works just fine [unless you're using Windoz, in which case God hates
>>>>> you**].
>>>>>
>>>>> For more refined control (if you know where your external contacts are
>>>>> coming from) ...
>>>>>
>>>>> Consider setting up ACLs (nailing down the IP address ranges from
>>>>> which you'll accept incalls) in autoload/acl.conf.xml -- the "domains"
>>>>> definition there is one place to add in your external correspondents.
>>>>>
>>>>> Also, consider using cidr= parameters in your directory folder for
>>>>> each of your users (if they will only attempt to register or place calls
>>>>> from given address ranges).
>>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>>
>>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and
>>>>> take it from there.
>>>>>
>>>>> BTW, you WILL be confused by setting explicit ACLs on registration --
>>>>> leave that one commented out until you know what it actually does, as it's
>>>>> probably not what you expect. Several strong cups of coffee and protracted
>>>>> meditation may help.
>>>>>
>>>>> Main message:
>>>>> -- Immediately - fix the passwords so they're not easy to guess [as
>>>>> the bad guys *will* try again and again until they get it right].
>>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>>> proposed. <======= MOST IMPORTANT
>>>>> -- lose the belief that firewalls are going to help protect an
>>>>> Internet-listening server as, logically, they can't
>>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>>> logfile, and wonder how you got away with it for so long.
>>>>>
>>>>> all the best,
>>>>>   Lawrence
>>>>> ** There was apparently a talk on how Windows users could get
>>>>> something close to a fail2ban-style setup (IIRC, it was on the weekly conf
>>>>> call a while back)
>>>>>
>>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>>>>> > I really think that people give way too much importance to firewalls,
>>>>> > specially stateless ones, blocking ports isn't going to do much for
>>>>> you
>>>>> > unless you are trying to hide vulnerable services behind it.
>>>>> >
>>>>> > They used the extension 1000 to make the calls so I would say:
>>>>> activate
>>>>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>>>>> > passwords.
>>>>> >
>>>>> > If you want to go further you can use a stateful firewall limiting
>>>>> > connections and setup a IDS(recommend snort)
>>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>>>>> >
>>>>> >>
>>>>> >> Hey All,
>>>>> >>
>>>>> >>
>>>>> >> I just got an email from Frontier that there were several attempts
>>>>> to
>>>>> >> make international calls.
>>>>> >>
>>>>> >>
>>>>> >> I checked the log file and verified that somehow someone was able
>>>>> to get
>>>>> >> access to FS from the internet.
>>>>> >>
>>>>> >>
>>>>> >> here is a sample of the log
>>>>> >>
>>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>> Ring-Ready
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>> switch_ivr_originate.c:519
>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>> Pre-Answer
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>> switch_ivr_originate.c:3303
>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>>> 16:30:29.916821
>>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>> Ring-Ready
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>> switch_ivr_originate.c:519
>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>> Pre-Answer
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>> switch_ivr_originate.c:3303
>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>>> >>
>>>>> >>
>>>>> >> At this point I'm at a loss how this is happening as I have multiple
>>>>> >> firewalls in place that limit port access.
>>>>> >>
>>>>> >> Can someone provide a few pointers on how to better secure FS
>>>>> running on
>>>>> >> Linux systems?
>>>>> >>
>>>>> >>
>>>>> >> thanks
>>>>> >>
>>>>> >>
>>>>> >> --
>>>>> >> -
>>>>> >> -
>>>>> >> -    Best Regards,
>>>>> >> -
>>>>> >> -            Todd Bailey
>>>>> >> -
>>>>> >> -
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> _________________________________________________________________________
>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>> >> consulting at freeswitch.org
>>>>> >> http://www.freeswitchsolutions.com
>>>>> >>
>>>>> >> 
>>>>> >> 
>>>>> >>
>>>>> >> Official FreeSWITCH Sites
>>>>> >> http://www.freeswitch.org
>>>>> >> http://wiki.freeswitch.org
>>>>> >> http://www.cluecon.com
>>>>> >>
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > 
>>>>> > 
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://wiki.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bigx333 at gmail.com  Thu Sep 27 23:53:44 2012
From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado)
Date: Thu, 27 Sep 2012 21:53:44 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
	<CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
Message-ID: <CAPt4n6--FiVGgrddPwqBSC2oeHOUrHS4JafjPPm+-o9m7D53yw@mail.gmail.com>

http://nmap.org/book/man-version-detection.html

The rabbit hole is much deeper than you think.
On Sep 27, 2012 9:29 PM, "BookBag" <asaad2 at gmail.com> wrote:

> when nmap finds a port open, it looks in its database of what protocol is
> likely to be running on that port. It doesnt actually test if the standard
> protocol is running on that port.
>
>
>
>
>
> On Thu, Sep 27, 2012 at 12:11 PM, Nelson Camargo <bigx333 at gmail.com>wrote:
>
>> Ever heard about nmap? lol
>> On 27 Sep 2012, at 5:52 PM, BookBag wrote:
>>
>> How will they know what protocol I'm running on that port?
>> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>
>>> This is classic wardialing and is very common. Don't worry, your port
>>> change won't slow down people who really want to get in ;)
>>>
>>>
>>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>>
>>>> I had the same issue. There are hackers continuously scanning public
>>>> ip's for known ports then trying to register devices using the default
>>>> extensions and passwords "1234". After noticing this in my logs I just
>>>> changed the default external sip port from 5080 to something else.
>>>>
>>>> Security through obscurity if you will.
>>>> P.S. I was also using fail2ban
>>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>>> wrote:
>>>>
>>>>> Hi There,
>>>>>  welcome to our world; hope it didn't cost too much.
>>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>>> them.
>>>>> I'd guess that this particular bunch are coming from IP addresses
>>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>>> appeared to originate.
>>>>>
>>>>> 1st rule of fight club: Firewalls are no use for a server that is
>>>>> going to listen for requests from the Internet and allow authenticated
>>>>> calls to be placed from any IP address.
>>>>>
>>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up
>>>>> and works just fine [unless you're using Windoz, in which case God hates
>>>>> you**].
>>>>>
>>>>> For more refined control (if you know where your external contacts are
>>>>> coming from) ...
>>>>>
>>>>> Consider setting up ACLs (nailing down the IP address ranges from
>>>>> which you'll accept incalls) in autoload/acl.conf.xml -- the "domains"
>>>>> definition there is one place to add in your external correspondents.
>>>>>
>>>>> Also, consider using cidr= parameters in your directory folder for
>>>>> each of your users (if they will only attempt to register or place calls
>>>>> from given address ranges).
>>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>>
>>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and
>>>>> take it from there.
>>>>>
>>>>> BTW, you WILL be confused by setting explicit ACLs on registration --
>>>>> leave that one commented out until you know what it actually does, as it's
>>>>> probably not what you expect. Several strong cups of coffee and protracted
>>>>> meditation may help.
>>>>>
>>>>> Main message:
>>>>> -- Immediately - fix the passwords so they're not easy to guess [as
>>>>> the bad guys *will* try again and again until they get it right].
>>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>>> proposed. <======= MOST IMPORTANT
>>>>> -- lose the belief that firewalls are going to help protect an
>>>>> Internet-listening server as, logically, they can't
>>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>>> logfile, and wonder how you got away with it for so long.
>>>>>
>>>>> all the best,
>>>>>   Lawrence
>>>>> ** There was apparently a talk on how Windows users could get
>>>>> something close to a fail2ban-style setup (IIRC, it was on the weekly conf
>>>>> call a while back)
>>>>>
>>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado wrote:
>>>>> > I really think that people give way too much importance to firewalls,
>>>>> > specially stateless ones, blocking ports isn't going to do much for
>>>>> you
>>>>> > unless you are trying to hide vulnerable services behind it.
>>>>> >
>>>>> > They used the extension 1000 to make the calls so I would say:
>>>>> activate
>>>>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>>>>> > passwords.
>>>>> >
>>>>> > If you want to go further you can use a stateful firewall limiting
>>>>> > connections and setup a IDS(recommend snort)
>>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net> wrote:
>>>>> >
>>>>> >>
>>>>> >> Hey All,
>>>>> >>
>>>>> >>
>>>>> >> I just got an email from Frontier that there were several attempts
>>>>> to
>>>>> >> make international calls.
>>>>> >>
>>>>> >>
>>>>> >> I checked the log file and verified that somehow someone was able
>>>>> to get
>>>>> >> access to FS from the internet.
>>>>> >>
>>>>> >>
>>>>> >> here is a sample of the log
>>>>> >>
>>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>> Ring-Ready
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>> switch_ivr_originate.c:519
>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>> Pre-Answer
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>> switch_ivr_originate.c:3303
>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>>> 16:30:29.916821
>>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>> Ring-Ready
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>> switch_ivr_originate.c:519
>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>> Pre-Answer
>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>> switch_ivr_originate.c:3303
>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>>> >>
>>>>> >>
>>>>> >> At this point I'm at a loss how this is happening as I have multiple
>>>>> >> firewalls in place that limit port access.
>>>>> >>
>>>>> >> Can someone provide a few pointers on how to better secure FS
>>>>> running on
>>>>> >> Linux systems?
>>>>> >>
>>>>> >>
>>>>> >> thanks
>>>>> >>
>>>>> >>
>>>>> >> --
>>>>> >> -
>>>>> >> -
>>>>> >> -    Best Regards,
>>>>> >> -
>>>>> >> -            Todd Bailey
>>>>> >> -
>>>>> >> -
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> _________________________________________________________________________
>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>> >> consulting at freeswitch.org
>>>>> >> http://www.freeswitchsolutions.com
>>>>> >>
>>>>> >> 
>>>>> >> 
>>>>> >>
>>>>> >> Official FreeSWITCH Sites
>>>>> >> http://www.freeswitch.org
>>>>> >> http://wiki.freeswitch.org
>>>>> >> http://www.cluecon.com
>>>>> >>
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > 
>>>>> > 
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://wiki.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 27 23:54:38 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 27 Sep 2012 15:54:38 -0400
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <CC8A0C72.54439%krice@freeswitch.org>
References: <CC8A0C72.54439%krice@freeswitch.org>
Message-ID: <EDBB9235-D08F-4FEC-A577-1D516FE6AA8F@kavun.ch>

Hey Ken,

I tried pass_rfc2833 with no noticeable change in the delay. It seemed to have made it less accurate though, especially in fast speed sequences.

Can I debug this further and how?

Thanks!
On Sep 27, 2012, at 3:00 PM, Ken Rice <krice at freeswitch.org> wrote:

> There can be a delay of DTMF in and DTMF out if you are sending long DTMFs
> using 2833, FreeSWITCH gets the entire DMTF and duration then regenerates
> it...
> 
> If you don't need to interpret the DTMF you can set a variable to make it
> just pass the DTMF through untouched... But this has its own set of caveats
> (ie: if whatever is sending you DTMF is broken it just pass broken 2833
> DTMF)
> 
> See http://wiki.freeswitch.org/wiki/Variable_pass_rfc2833
> 
> 
> 
> On 9/27/12 1:49 PM, "Emrah" <lists at kavun.ch> wrote:
> 
>> MC, the issue does not happen with inband DTMF and there is no delay!
>> 
>> Any idea on how to debug this further? I can't use inband continuously.
>> 
>> Thanks!
>> Emrah
>> 
>> On Sep 27, 2012, at 12:46 PM, Emrah <lists at kavun.ch> wrote:
>> 
>>> Never tried with inband DTMFs. Will check.
>>> 
>>> Thanks!
>>> On Sep 27, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> 
>>>> Does it happen whether you use RFC2833 or inband DTMFs? Just curious.
>>>> -MC
>>>> 
>>>> On Wed, Sep 26, 2012 at 3:44 PM, Emrah <lists at kavun.ch> wrote:
>>>> Yes I did.
>>>> BTW, the example in the Wiki contradicts the inline documentation in
>>>> switch.xml.
>>>>   <!--
>>>>        The min-dtmf-duration specifies the minimum DTMF duration to use on
>>>>        outgoing events. Events shorter than this will be increased in
>>>> duration
>>>>        to match min_dtmf_duration. You cannot configure a dtmf duration on
>>>> a
>>>>        profile that is less than this setting. You may increase this value,
>>>>        but cannot set it lower than 400. This value cannot exceed
>>>>        max-dtmf-duration. -->
>>>> The Wiki shows an example with the value at 100.
>>>> 
>>>> I tried increasing and decreasing it to no avail, it does not seem to
>>>> interfere with anything I can measure with my ear. :P
>>>> On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com>
>>>> wrote:
>>>> 
>>>>> You tried this:
>>>>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
>>>>> 
>>>>> On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
>>>>> Hi guys,
>>>>> 
>>>>> I am comparing this with an Asterisk and FreeSWITCH installation, using the
>>>>> same route, same codecs, same carrier, same phones and same servers? :P
>>>>> I experience a delay when pressing DTMFs on the line that uses FreeSWITCH.
>>>>> I am estimating the delay to be around 500 ms.
>>>>> 
>>>>> What are the settings I can fine tune to avoid this?
>>>>> 
>>>>> All the 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
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
> 
> -- 
> 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



From lists at telefaks.de  Fri Sep 28 01:24:49 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 27 Sep 2012 23:24:49 +0200
Subject: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is registered
	at another freeswitch.
Message-ID: <5064C421.9010206@telefaks.de>

Hello,

I have 2 Freeswitch boxes.
Box 2 is registering as number 400 via VPN to box 1 (like a regular
phone). I can see in the sip_registrations database that the phone is
registered. And Box2 shows "
   vpn::sip1.mydomain.com       gateway                     
sip:400 at 10.8.0.1:5075      REGED
So at the first glance everything seems ok.

However in my vpn profile
   sofia status profile internalvpn
the Freeswitch box2 is not shown as a registration entry.

I think this is the reason why we receive the following when we try to
call the box 2 (number 400):
EXECUTE sofia/internal/200 at sip1.mydomain.com
bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
2012-09-27 23:09:58.149196 [DEBUG] switch_channel.c:1062
sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
[dialed_ext]=[400] to event
2012-09-27 23:09:58.149196 [DEBUG] switch_ivr_originate.c:1961 Parsing
global variables
2012-09-27 23:09:58.149196 [DEBUG] switch_event.c:1478 Parsing variable
[sip_invite_domain]=[sip1.mydomain.com]
2012-09-27 23:09:58.169252 [DEBUG] switch_core_session.c:924 Send signal
sofia/internal/200 at sip1.mydomain.com [BREAK]
2012-09-27 23:09:58.269299 [DEBUG] switch_channel.c:1062
sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
[dialed_ext]=[400] to event
2012-09-27 23:09:58.269299 [DEBUG] switch_ivr_originate.c:1961 Parsing
global variables
2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
[presence_id]=[400 at sip1.mydomain.com]
2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
[transfer_fallback_extension]=[400]
2012-09-27 23:09:58.269299 [DEBUG] sofia.c:6040 Channel
sofia/internal/200 at sip1.mydomain.com entering state [early][180]
2012-09-27 23:09:58.269299 [NOTICE] switch_ivr_originate.c:2544 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]


freeswitch at internal> sofia_contact 400
error/user_not_registered
freeswitch at internal> sofia_contact 200
sofia/internal/sip:200 at 192.168.178.105:2048;line=1vz0u23u;fs_nat=yes;fs_path=sip%3A200%40217.24.xx.xxx%3A20393%3Bline%3D1vz0u23u
So 200 seems ok, 400 not.

When I look at the contact field in the sip_registrations table, I see
"user"<sip:gw+sip1.mydomain.com at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com>
so this does not reflect the number 400.

Here is my gateway config from box2 which registers at box1
(sip1.mydomain.com)
# cat sip1.xml
<include>
  <gateway name="sip1.mydomain.com">
    <param name="from-domain" value="sip1.mydomain.com"/>
    <param name="username" value="400"/>
    <param name="from-user" value="400"/>
    <param name="extension" value="400"/>
    <param name="realm" value="sip1.mydomain.com"/>
    <!-- 5075 registers to the internal VPN profile -->
    <param name="proxy" value="10.8.0.1:5075"/>
    <param name="password" value="xxxxxxxx"/>
    <param name="expire-seconds" value="3600"/>
    <param name="register" value="true"/>
    <param name="register-transport" value="udp"/>
    <param name="retry-seconds" value="20"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>


My question is: How can I change the gateway definition, so that box2
shows up as a regular registration with number 400? Or am I missing
something different?

-- 
With kind regards
Peter Steinbach 

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




From msc at freeswitch.org  Fri Sep 28 01:47:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Sep 2012 14:47:31 -0700
Subject: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is
 registered at another freeswitch.
In-Reply-To: <5064C421.9010206@telefaks.de>
References: <5064C421.9010206@telefaks.de>
Message-ID: <CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>

Just for kicks, try this:

sofia_contact */400

I'm curious about something.
-MC

On Thu, Sep 27, 2012 at 2:24 PM, Peter Steinbach <lists at telefaks.de> wrote:

> Hello,
>
> I have 2 Freeswitch boxes.
> Box 2 is registering as number 400 via VPN to box 1 (like a regular
> phone). I can see in the sip_registrations database that the phone is
> registered. And Box2 shows "
>    vpn::sip1.mydomain.com       gateway
> sip:400 at 10.8.0.1:5075      REGED
> So at the first glance everything seems ok.
>
> However in my vpn profile
>    sofia status profile internalvpn
> the Freeswitch box2 is not shown as a registration entry.
>
> I think this is the reason why we receive the following when we try to
> call the box 2 (number 400):
> EXECUTE sofia/internal/200 at sip1.mydomain.com
> bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
> 2012-09-27 23:09:58.149196 [DEBUG] switch_channel.c:1062
> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
> [dialed_ext]=[400] to event
> 2012-09-27 23:09:58.149196 [DEBUG] switch_ivr_originate.c:1961 Parsing
> global variables
> 2012-09-27 23:09:58.149196 [DEBUG] switch_event.c:1478 Parsing variable
> [sip_invite_domain]=[sip1.mydomain.com]
> 2012-09-27 23:09:58.169252 [DEBUG] switch_core_session.c:924 Send signal
> sofia/internal/200 at sip1.mydomain.com [BREAK]
> 2012-09-27 23:09:58.269299 [DEBUG] switch_channel.c:1062
> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
> [dialed_ext]=[400] to event
> 2012-09-27 23:09:58.269299 [DEBUG] switch_ivr_originate.c:1961 Parsing
> global variables
> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
> [presence_id]=[400 at sip1.mydomain.com]
> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
> [transfer_fallback_extension]=[400]
> 2012-09-27 23:09:58.269299 [DEBUG] sofia.c:6040 Channel
> sofia/internal/200 at sip1.mydomain.com entering state [early][180]
> 2012-09-27 23:09:58.269299 [NOTICE] switch_ivr_originate.c:2544 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>
>
> freeswitch at internal> sofia_contact 400
> error/user_not_registered
> freeswitch at internal> sofia_contact 200
> sofia/internal/sip:200 at 192.168.178.105:2048
> ;line=1vz0u23u;fs_nat=yes;fs_path=sip%3A200%40217.24.xx.xxx%3A20393%3Bline%3D1vz0u23u
> So 200 seems ok, 400 not.
>
> When I look at the contact field in the sip_registrations table, I see
> "user"<sip:gw+sip1.mydomain.com at 10.8.0.42:5080;transport=udp;gw=
> sip1.mydomain.com>
> so this does not reflect the number 400.
>
> Here is my gateway config from box2 which registers at box1
> (sip1.mydomain.com)
> # cat sip1.xml
> <include>
>   <gateway name="sip1.mydomain.com">
>     <param name="from-domain" value="sip1.mydomain.com"/>
>     <param name="username" value="400"/>
>     <param name="from-user" value="400"/>
>     <param name="extension" value="400"/>
>     <param name="realm" value="sip1.mydomain.com"/>
>     <!-- 5075 registers to the internal VPN profile -->
>     <param name="proxy" value="10.8.0.1:5075"/>
>     <param name="password" value="xxxxxxxx"/>
>     <param name="expire-seconds" value="3600"/>
>     <param name="register" value="true"/>
>     <param name="register-transport" value="udp"/>
>     <param name="retry-seconds" value="20"/>
>     <param name="caller-id-in-from" value="true"/>
>   </gateway>
> </include>
>
>
> My question is: How can I change the gateway definition, so that box2
> shows up as a regular registration with number 400? Or am I missing
> something different?
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
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From msc at freeswitch.org  Fri Sep 28 02:11:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Sep 2012 15:11:32 -0700
Subject: [Freeswitch-users] Fwd: [Freeswitch-dev] :how to let the
 version 2.1.3 as well as the latest (from git) work for the 0911 and 5001
 dialplan? there is no problem with version 1.06
In-Reply-To: <CAA0cQJfQQSjAy2MVZYK9zgp7EixYc2X-E2N=aczn+tuVKC=+Ag@mail.gmail.com>
References: <CAA0cQJc_Dm5W6GbOWBW7hMQB_xSM72Kpk2_y2A5dcO6d10gwvw@mail.gmail.com>
	<CC896CDC.543F5%krice@freeswitch.org>
	<CAA0cQJepzETnBCiF01TBXRPy8yf8EgGAEkyn1P5A1UmhRP8rkA@mail.gmail.com>
	<CAA0cQJeUsJGBQ-LfwxdwWxUkvR=Ffum0x73VwJd5hCoT1tkcng@mail.gmail.com>
	<CAA0cQJfQQSjAy2MVZYK9zgp7EixYc2X-E2N=aczn+tuVKC=+Ag@mail.gmail.com>
Message-ID: <CAKzWOxWq_o39KMhzJ=5SiKB2oFnwG+tHKvq5hW7+VKsyfL8iSA@mail.gmail.com>

zhisun,

I was able to replicate this on stable 1.2.3 with a clean install. Can you
please open a jira on this?
Thanks,
MC

On Thu, Sep 27, 2012 at 2:03 AM, zhi sun <sunzhimailbox at gmail.com> wrote:

> just forward this email to users group, in case anyone else have
> encountered the same problem!
>
> thanks for your help
> -zhisun
>
>
> ---------- Forwarded message ----------
> From: zhi sun <sunzhimailbox at gmail.com>
> Date: 2012/9/27
> Subject: Re: [Freeswitch-dev] :how to let the version 2.1.3 as well as the
> latest (from git) work for the 0911 and 5001 dialplan? there is no problem
> with version 1.06
> To: freeswitch-dev at lists.freeswitch.org
>
>
> what i am concerning is how to make the version 1.2.3, as well as the
> latest git version works for 0911 and 5001 in default.xml, although the
> version 1.0.6 only works partly on this issue.
>
> this problem is very easy to reproduce!!!!
>
>
> 2012/9/27 zhi sun <sunzhimailbox at gmail.com>
>
>> thank ken for your response.
>>
>> how to update the configs? do you mean the conf folder? i am new to
>> freeswitch.
>>
>> what i have done is just use the conf created by make install command.
>>
>> since i set different installation path by using ./configure
>> --prefix=/my/path, either ver 1.2.3 or ver 1.0.6 has different conf folder
>> i think.
>>
>> at least, i believe that the config with the version 1.2.3 is shipped
>> with ver 1.2.3.
>>
>> thanks,
>> -zhisun
>>
>>
>> 2012/9/27 Ken Rice <krice at freeswitch.org>
>>
>>>  Did you update your configs? 1.0.6 is so old and has more issues then
>>> I can count... And configs have been update to resolve problems...
>>>
>>>
>>> On 9/27/12 2:02 AM, "zhi sun" <sunzhimailbox at gmail.com> wrote:
>>>
>>> further testing results:
>>>
>>> - for version 1.0.6, the 0911 works, but 5001 doesn't.
>>>
>>> - for version 1.2.3 and 1.3, both 0911 and 5001 doesn't work.
>>>
>>> the following are logs for 0911, version 1.2.3
>>>
>>> the 1000,1001,1002,1003 should be called!!!!
>>>
>>> ==============================================
>>> 2012-09-27 14:58:47.233466 [NOTICE] switch_channel.c:951 New Channel
>>> sofia/internal/1001 at 192.168.0.100 [c8c8f364-0870-11e2-bd7f-df6c04573b9b]
>>> 2012-09-27 14:58:47.273467 [INFO] mod_dialplan_xml.c:485 Processing 1001
>>> <1001>->0911 in context default
>>> 2012-09-27 14:58:47.293472 [INFO] switch_core_session.c:2392 Sending
>>> early media
>>> 2012-09-27 14:58:47.293472 [NOTICE] sofia_glue.c:4226 Pre-Answer
>>> sofia/internal/1001 at 192.168.0.100!
>>> 2012-09-27 14:58:47.293472 [NOTICE] mod_conference.c:7211 Channel [
>>> sofia/internal/1001 at 192.168.0.100] has been answered
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8315 using channel
>>> sound prefix:
>>> /home/sunzhi/sunzhi/freeswitch/freeswitch-1.2.3/rel/sounds/en/us/callie
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '0' to 'mute'
>>> 2012-09-27 14:58:47.293472 [INFO] switch_ivr_async.c:194 Digit parser
>>> mod_conference: Setting realm to 'conf'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '*' to 'deaf mute'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '9' to 'energy up'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '8' to 'energy equ'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '7' to 'energy dn'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '3' to 'vol talk up'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '2' to 'vol talk zero'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '1' to 'vol talk dn'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '6' to 'vol listen up'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '5' to 'vol listen zero'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '4' to 'vol listen dn'
>>> 2012-09-27 14:58:47.293472 [INFO] mod_conference.c:8869
>>> sofia/internal/1001 at 192.168.0.100 binding '#' to 'hangup'
>>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1001 at 192.168.0.100]error
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1000 at 192.168.0.100]sofia
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1002 at 192.168.0.100]sofia
>>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1002 at 192.168.0.100]sofia]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1001 at 192.168.0.100]error]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1003 at 192.168.0.100]error
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1003 at 192.168.0.100]error]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1000 at 192.168.0.100]sofia]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 14:58:47.293472 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1004 at 192.168.0.100]error
>>> 2012-09-27 14:58:47.293472 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1004 at 192.168.0.100]error]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 14:58:47.293472 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 14:58:47.293472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> ========================================================
>>>
>>>
>>> 2012/9/27 zhi sun <sunzhimailbox at gmail.com>
>>>
>>> in addition to previous email, i also try the following:
>>>
>>> - add a new diaplan (5002) similar to 5001 in default.xml
>>>
>>>     <extension name="dynamic_conference">
>>>       <condition field="destination_number" expression="^5002$">
>>>     <action application="conference" data="bridge:mydynaconf:
>>> sofia/${use_profile}/1002 at 192.168.0.100"/>
>>>       </condition>
>>>     </extension>
>>>
>>> - reloadxml
>>>
>>> - then call 5002 from 1010 sip client,
>>>
>>> - the same problem happens
>>>
>>> PS: it works fine on version 1.06
>>>
>>> ==========================================================
>>> freeswitch at mydev.mydomain.com> reloadxml
>>>
>>> +OK [Success]
>>>
>>> 2012-09-27 09:46:46.613468 [INFO] mod_enum.c:871 ENUM Reloaded
>>> 2012-09-27 09:46:46.613468 [INFO] switch_time.c:1163 Timezone reloaded
>>> 530 definitions
>>> freeswitch at mydev.mydomain.com> 2012-09-27 09:47:19.413468 [NOTICE]
>>> switch_channel.c:951 New Channel sofia/internal/1010 at 192.168.0.100[45f8ccbe-0845-11e2-b58c-8d63071eb0f5]
>>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_NEW
>>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_state_machine.c:416 (
>>> sofia/internal/1010 at 192.168.0.100) State NEW
>>> 2012-09-27 09:47:19.413468 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.433469 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>>> by acl "domains". Falling back to Digest auth.
>>> 2012-09-27 09:47:19.433469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.433469 [DEBUG] sofia.c:1728 detaching session
>>> 45f8ccbe-0845-11e2-b58c-8d63071eb0f5
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:1820 Re-attaching to session
>>> 45f8ccbe-0845-11e2-b58c-8d63071eb0f5
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>>> by acl "domains". Falling back to Digest auth.
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1010 at 192.168.0.100 entering state [received][100]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6293 Remote SDP:
>>> v=0
>>> o=1010 3519 3519 IN IP4 192.168.0.10
>>> s=Talk
>>> c=IN IP4 192.168.0.10
>>> t=0 0
>>> m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
>>> a=rtpmap:112 speex/32000
>>> a=fmtp:112 vbr=on
>>> a=rtpmap:111 speex/16000
>>> a=fmtp:111 vbr=on
>>> a=rtpmap:110 speex/8000
>>> a=fmtp:110 vbr=on
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-11
>>>
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6506 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/1010 at 192.168.0.100) State INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:86
>>> sofia/internal/1010 at 192.168.0.100 SOFIA INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:126 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_INIT -> CS_ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/1010 at 192.168.0.100) State INIT going to sleep
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1964 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change DOWN -> RINGING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/1010 at 192.168.0.100) State ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:149
>>> sofia/internal/1010 at 192.168.0.100 SOFIA ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:117
>>> sofia/internal/1010 at 192.168.0.100 Standard ROUTING
>>> 2012-09-27 09:47:19.453469 [INFO] mod_dialplan_xml.c:485 Processing 1010
>>> <1010>->5002 in context default
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->unloop]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS) [unloop]
>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [unloop]
>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->tod_example] continue=true
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/Time Match (PASS)
>>> [tod_example] break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action set(open=true)
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->holiday_example] continue=true
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/TimeMatch (FAIL)
>>> [holiday_example] break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->global-intercept] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [global-intercept] destination_number(5002) =~ /^886$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group-intercept] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group-intercept] destination_number(5002) =~ /^\*8$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->intercept-ext] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [intercept-ext] destination_number(5002) =~ /^\*\*(\d+)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->redial]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [redial]
>>> destination_number(5002) =~ /^(redial|870)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->global]
>>> continue=true
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>>> ${call_debug}(false) =~ /^true$/ break=never
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>>> ${sip_has_crypto}() =~
>>> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Absolute Condition [global]
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>>
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> hash(insert/${domain_name}-last_dial/global/${uuid})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->snom-demo-2] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-2]
>>> destination_number(5002) =~ /^9001$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->snom-demo-1] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-1]
>>> destination_number(5002) =~ /^9000$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->eavesdrop] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>>> destination_number(5002) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->eavesdrop] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>>> destination_number(5002) =~ /^779$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->call_return] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [call_return]
>>> destination_number(5002) =~ /^\*69$|^869$|^lcr$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->del-group] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [del-group]
>>> destination_number(5002) =~ /^80(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->add-group] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [add-group]
>>> destination_number(5002) =~ /^81(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->call-group-simo] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [call-group-simo] destination_number(5002) =~ /^82(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->call-group-order] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [call-group-order] destination_number(5002) =~ /^83(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->extension-intercom] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [extension-intercom] destination_number(5002) =~ /^8(10[01][0-9])$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->Local_Extension] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [Local_Extension] destination_number(5002) =~ /^(10[01][0-9])$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->Local_Extension_Skinny] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [Local_Extension_Skinny] destination_number(5002) =~ /^(11[01][0-9])$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group_dial_sales] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group_dial_sales] destination_number(5002) =~ /^2000$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group_dial_support] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group_dial_support] destination_number(5002) =~ /^2001$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group_dial_billing] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group_dial_billing] destination_number(5002) =~ /^2002$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->operator]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [operator]
>>> destination_number(5002) =~ /^(operator|0)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->vmain]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [vmain]
>>> destination_number(5002) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->sip_uri]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [sip_uri]
>>> destination_number(5002) =~ /^sip:(.*)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->nb_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [nb_conferences] destination_number(5002) =~ /^(30\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->wb_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [wb_conferences] destination_number(5002) =~ /^(31\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->uwb_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [uwb_conferences] destination_number(5002) =~ /^(32\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->cdquality_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [cdquality_conferences] destination_number(5002) =~ /^(33\d{2})$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->freeswitch_public_conf_via_sip] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [freeswitch_public_conf_via_sip] destination_number(5002) =~
>>> /^9(888|8888|1616|3232)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->mad_boss_intercom] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [mad_boss_intercom] destination_number(5002) =~ /^0911$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->mad_boss_intercom] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [mad_boss_intercom] destination_number(5002) =~ /^0912$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->mad_boss]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [mad_boss]
>>> destination_number(5002) =~ /^0913$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->ivr_demo]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [ivr_demo]
>>> destination_number(5002) =~ /^5000$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->dynamic_conference] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [dynamic_conference] destination_number(5002) =~ /^5001$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->dynamic_conference] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS)
>>> [dynamic_conference] destination_number(5002) =~ /^5002$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> conference(bridge:mydynaconf:sofia/${use_profile}/1002 at 192.168.0.100)
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:167 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_ROUTING -> CS_EXECUTE
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/1010 at 192.168.0.100) State ROUTING going to sleep
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_EXECUTE
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:453 (
>>> sofia/internal/1010 at 192.168.0.100) State EXECUTE
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:242
>>> sofia/internal/1010 at 192.168.0.100 SOFIA EXECUTE
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:209
>>> sofia/internal/1010 at 192.168.0.100 Standard EXECUTE
>>> EXECUTE sofia/internal/1010 at 192.168.0.100 set(open=true)
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET [open]=[true]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-spymap/1010/45f8ccbe-0845-11e2-b58c-8d63071eb0f5)
>>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/1010/5002)
>>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/global/45f8ccbe-0845-11e2-b58c-8d63071eb0f5)
>>> EXECUTE sofia/internal/1010 at 192.168.0.100 export(RFC2822_DATE=Thu, 27
>>> Sep 2012 09:47:19 +0800)
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1118 EXPORT
>>> (export_vars) [RFC2822_DATE]=[Thu, 27 Sep 2012 09:47:19 +0800]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:2390
>>> Application conference Requires media! pre_answering channel
>>> sofia/internal/1010 at 192.168.0.100
>>> 2012-09-27 09:47:19.453469 [INFO] switch_core_session.c:2392 Sending
>>> early media
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3077 Set Codec
>>> sofia/internal/1010 at 192.168.0.100 GSM/8000 20 ms 160 samples 13200 bits
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_codec.c:111
>>> sofia/internal/1010 at 192.168.0.100 Original read codec set to GSM:3
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:5219 Set 2833 dtmf
>>> send/recv payload to 101
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3327 AUDIO RTP [
>>> sofia/internal/1010 at 192.168.0.100] 192.168.0.100 port 31092 ->
>>> 192.168.0.10 port 7078 codec: 3 ms: 20
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_rtp.c:1927 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3591 Set 2833 dtmf send
>>> payload to 101
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3597 Set 2833 dtmf
>>> receive payload to 101
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:3624
>>> sofia/internal/1010 at 192.168.0.100 Set rtp dtmf delay to 40
>>> 2012-09-27 09:47:19.453469 [NOTICE] sofia_glue.c:4226 Pre-Answer
>>> sofia/internal/1010 at 192.168.0.100!
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:3092 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change RINGING -> EARLY
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:2730 Ring SDP:
>>> v=0
>>> o=FreeSWITCH 1348679347 1348679348 IN IP4 192.168.0.100
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.0.100
>>> t=0 0
>>> m=audio 31092 RTP/AVP 3 101
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:759 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100 conference(bridge:mydynaconf:
>>> sofia/internal/1002 at 192.168.0.100)
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1010 at 192.168.0.100 entering state [early][183]
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:836 Local SDP
>>> sofia/internal/1010 at 192.168.0.100:
>>> v=0
>>> o=FreeSWITCH 1348679347 1348679349 IN IP4 192.168.0.100
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.0.100
>>> t=0 0
>>> m=audio 31092 RTP/AVP 3 101
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:759 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:3351 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change EARLY -> ACTIVE
>>> 2012-09-27 09:47:19.453469 [NOTICE] mod_conference.c:7211 Channel [
>>> sofia/internal/1010 at 192.168.0.100] has been answered
>>> 2012-09-27 09:47:19.453469 [INFO] mod_conference.c:8315 using channel
>>> sound prefix:
>>> /home/sunzhi/sunzhi/freeswitch/freeswitch-1.2.3/rel/sounds/en/us/callie
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_conference.c:1922 Setup timer
>>> success interval: 20  samples: 160
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1072
>>> sofia/internal/1010 at 192.168.0.100 EXPORTING[export_vars]
>>> [RFC2822_DATE]=[Thu, 27 Sep 2012 09:47:19 +0800] to event
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:47:19.453469 [NOTICE] switch_channel.c:951 New Channel
>>> sofia/internal/1002 at 192.168.0.100 [46000434-0845-11e2-b598-8d63071eb0f5]
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:4879 (
>>> sofia/internal/1002 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:4954 [zrtp_passthru]
>>> Setting a-leg inherit_codec=true
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:4957 [zrtp_passthru]
>>> Setting b-leg absolute_codec_string='GSM at 8000h@20i at 13200b,PCMU at 8000h
>>> @20i at 64000b,PCMA at 8000h@20i at 64000b'
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1010 at 192.168.0.100 entering state [completed][200]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/1002 at 192.168.0.100) State INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:86
>>> sofia/internal/1002 at 192.168.0.100 SOFIA INIT
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia_glue.c:2637 Local SDP:
>>> v=0
>>> o=FreeSWITCH 1348687005 1348687006 IN IP4 192.168.0.100
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.0.100
>>> t=0 0
>>> m=audio 23434 RTP/AVP 3 0 8 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:126 (
>>> sofia/internal/1002 at 192.168.0.100) State Change CS_INIT -> CS_ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/1002 at 192.168.0.100) State INIT going to sleep
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_channel.c:1964 (
>>> sofia/internal/1002 at 192.168.0.100) Callstate Change DOWN -> RINGING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/1002 at 192.168.0.100) State ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] mod_sofia.c:149
>>> sofia/internal/1002 at 192.168.0.100 SOFIA ROUTING
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_ivr_originate.c:67 (
>>> sofia/internal/1002 at 192.168.0.100) State Change CS_ROUTING ->
>>> CS_CONSUME_MEDIA
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/1002 at 192.168.0.100) State ROUTING going to sleep
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_CONSUME_MEDIA
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:465 (
>>> sofia/internal/1002 at 192.168.0.100) State CONSUME_MEDIA
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:465 (
>>> sofia/internal/1002 at 192.168.0.100) State CONSUME_MEDIA going to sleep
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1002 at 192.168.0.100 entering state [calling][0]
>>> 2012-09-27 09:47:19.453469 [NOTICE] switch_channel.c:951 New Channel
>>> sofia/internal/0000000000 at 192.168.0.100[46006712-0845-11e2-b59c-8d63071eb0f5]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_NEW
>>> 2012-09-27 09:47:19.453469 [DEBUG] switch_core_state_machine.c:416 (
>>> sofia/internal/0000000000 at 192.168.0.100) State NEW
>>> 2012-09-27 09:47:19.473478 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:8412 IP 192.168.0.100
>>> Rejected by acl "domains". Falling back to Digest auth.
>>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/0000000000 at 192.168.0.100 entering state [received][100]
>>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:6293 Remote SDP:
>>> v=0
>>> o=FreeSWITCH 1348687005 1348687006 IN IP4 192.168.0.100
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.0.100
>>> t=0 0
>>> m=audio 23434 RTP/AVP 3 0 8 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:6506 (
>>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_INIT
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/0000000000 at 192.168.0.100) State INIT
>>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:86
>>> sofia/internal/0000000000 at 192.168.0.100 SOFIA INIT
>>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:126 (
>>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_INIT ->
>>> CS_ROUTING
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/0000000000 at 192.168.0.100) State INIT going to sleep
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_ROUTING
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_channel.c:1964 (
>>> sofia/internal/0000000000 at 192.168.0.100) Callstate Change DOWN ->
>>> RINGING
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/0000000000 at 192.168.0.100) State ROUTING
>>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:149
>>> sofia/internal/0000000000 at 192.168.0.100 SOFIA ROUTING
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:117
>>> sofia/internal/0000000000 at 192.168.0.100 Standard ROUTING
>>> 2012-09-27 09:47:19.493470 [INFO] mod_dialplan_xml.c:485 Processing
>>> FreeSWITCH <0000000000>->1002 in context public
>>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 parsing
>>> [public->unloop] continue=false
>>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 Regex (PASS) [unloop]
>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 Regex (PASS) [unloop]
>>> ${sip_looped_call}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/0000000000 at 192.168.0.100 Action
>>> deflect(${destination_number})
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:167 (
>>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_ROUTING ->
>>> CS_EXECUTE
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/0000000000 at 192.168.0.100) State ROUTING going to sleep
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_EXECUTE
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:453 (
>>> sofia/internal/0000000000 at 192.168.0.100) State EXECUTE
>>> 2012-09-27 09:47:19.493470 [DEBUG] mod_sofia.c:242
>>> sofia/internal/0000000000 at 192.168.0.100 SOFIA EXECUTE
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_state_machine.c:209
>>> sofia/internal/0000000000 at 192.168.0.100 Standard EXECUTE
>>> EXECUTE sofia/internal/0000000000 at 192.168.0.100 deflect(1002)
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] sofia.c:7308 Process REFER to [
>>> 1002 at 192.168.0.100]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_ivr.c:1742 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_EXECUTE -> CS_ROUTING
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:759 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [NOTICE] switch_ivr.c:1748 Transfer
>>> sofia/internal/1010 at 192.168.0.100 to XML[1002 at default]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.493470 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2950 (
>>> sofia/internal/1002 at 192.168.0.100) Callstate Change RINGING -> HANGUP
>>> 2012-09-27 09:47:19.513472 [NOTICE] switch_ivr_originate.c:3326 Hangup
>>> sofia/internal/1002 at 192.168.0.100 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2973 Send signal
>>> sofia/internal/1002 at 192.168.0.100 [KILL]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_HANGUP
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_ivr_originate.c:3502 Originate
>>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>> 2012-09-27 09:47:19.513472 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: ORIGINATOR_CANCEL
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2950 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change ACTIVE -> HANGUP
>>> 2012-09-27 09:47:19.513472 [NOTICE] mod_conference.c:6617 Hangup
>>> sofia/internal/1010 at 192.168.0.100 [CS_ROUTING] [ORIGINATOR_CANCEL]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/1002 at 192.168.0.100) State HANGUP
>>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:483 Channel
>>> sofia/internal/1002 at 192.168.0.100 hanging up, cause: ORIGINATOR_CANCEL
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_channel.c:2973 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [KILL]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:2553
>>> sofia/internal/1010 at 192.168.0.100 skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:453 (
>>> sofia/internal/1010 at 192.168.0.100) State EXECUTE going to sleep
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_HANGUP
>>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:542 Sending CANCEL to
>>> sofia/internal/1002 at 192.168.0.100
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:48
>>> sofia/internal/1002 at 192.168.0.100 Standard HANGUP, cause:
>>> ORIGINATOR_CANCEL
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/1002 at 192.168.0.100) State HANGUP going to sleep
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:429 (
>>> sofia/internal/1002 at 192.168.0.100) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_REPORTING
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/1002 at 192.168.0.100) State REPORTING
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:92
>>> sofia/internal/1002 at 192.168.0.100 Standard REPORTING, cause:
>>> ORIGINATOR_CANCEL
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/1002 at 192.168.0.100) State REPORTING going to sleep
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:423 (
>>> sofia/internal/1002 at 192.168.0.100) State Change CS_REPORTING ->
>>> CS_DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1002 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/1010 at 192.168.0.100) State HANGUP
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1415 Session 5 (
>>> sofia/internal/1002 at 192.168.0.100) Locked, Waiting on external entities
>>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:483 Channel
>>> sofia/internal/1010 at 192.168.0.100 hanging up, cause: ORIGINATOR_CANCEL
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:532 Sending BYE to
>>> sofia/internal/1010 at 192.168.0.100
>>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1433 Session 5
>>> (sofia/internal/1002 at 192.168.0.100) Ended
>>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1437 Close
>>> Channel sofia/internal/1002 at 192.168.0.100 [CS_DESTROY]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:48
>>> sofia/internal/1010 at 192.168.0.100 Standard HANGUP, cause:
>>> ORIGINATOR_CANCEL
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/1010 at 192.168.0.100) State HANGUP going to sleep
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:429 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_REPORTING
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/1010 at 192.168.0.100) State REPORTING
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:527 (
>>> sofia/internal/1002 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:92
>>> sofia/internal/1010 at 192.168.0.100 Standard REPORTING, cause:
>>> ORIGINATOR_CANCEL
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/1010 at 192.168.0.100) State REPORTING going to sleep
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:530 (
>>> sofia/internal/1002 at 192.168.0.100) Running State Change CS_DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/1002 at 192.168.0.100) State DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:376
>>> sofia/internal/1002 at 192.168.0.100 SOFIA DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:99
>>> sofia/internal/1002 at 192.168.0.100 Standard DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/1002 at 192.168.0.100) State DESTROY going to sleep
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:423 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_REPORTING ->
>>> CS_DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:1415 Session 4 (
>>> sofia/internal/1010 at 192.168.0.100) Locked, Waiting on external entities
>>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1433 Session 4
>>> (sofia/internal/1010 at 192.168.0.100) Ended
>>> 2012-09-27 09:47:19.513472 [NOTICE] switch_core_session.c:1437 Close
>>> Channel sofia/internal/1010 at 192.168.0.100 [CS_DESTROY]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:527 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:530 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/1010 at 192.168.0.100) State DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] mod_sofia.c:376
>>> sofia/internal/1010 at 192.168.0.100 SOFIA DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:99
>>> sofia/internal/1010 at 192.168.0.100 Standard DESTROY
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/1010 at 192.168.0.100) State DESTROY going to sleep
>>> 2012-09-27 09:47:19.513472 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.533467 [DEBUG] mod_conference.c:2461 Write Lock ON
>>> 2012-09-27 09:47:19.533467 [DEBUG] mod_conference.c:2464 Write Lock OFF
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_channel.c:2950 (
>>> sofia/internal/0000000000 at 192.168.0.100) Callstate Change RINGING ->
>>> HANGUP
>>> 2012-09-27 09:47:19.593467 [NOTICE] sofia.c:420 Hangup
>>> sofia/internal/0000000000 at 192.168.0.100 [CS_EXECUTE] [BLIND_TRANSFER]
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_channel.c:2973 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [KILL]
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:2553
>>> sofia/internal/0000000000 at 192.168.0.100 skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:453 (
>>> sofia/internal/0000000000 at 192.168.0.100) State EXECUTE going to sleep
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_HANGUP
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/0000000000 at 192.168.0.100) State HANGUP
>>> 2012-09-27 09:47:19.593467 [DEBUG] mod_sofia.c:483 Channel
>>> sofia/internal/0000000000 at 192.168.0.100 hanging up, cause:
>>> BLIND_TRANSFER
>>> 2012-09-27 09:47:19.593467 [DEBUG] mod_sofia.c:613 Responding to INVITE
>>> with: 480
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:48
>>> sofia/internal/0000000000 at 192.168.0.100 Standard HANGUP, cause:
>>> BLIND_TRANSFER
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/0000000000 at 192.168.0.100) State HANGUP going to sleep
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:429 (
>>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/0000000000 at 192.168.0.100) Running State Change
>>> CS_REPORTING
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/0000000000 at 192.168.0.100) State REPORTING
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:92
>>> sofia/internal/0000000000 at 192.168.0.100 Standard REPORTING, cause:
>>> BLIND_TRANSFER
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/0000000000 at 192.168.0.100) State REPORTING going to sleep
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:423 (
>>> sofia/internal/0000000000 at 192.168.0.100) State Change CS_REPORTING ->
>>> CS_DESTROY
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_session.c:1415 Session 6 (
>>> sofia/internal/0000000000 at 192.168.0.100) Locked, Waiting on external
>>> entities
>>> 2012-09-27 09:47:19.593467 [NOTICE] switch_core_session.c:1433 Session 6
>>> (sofia/internal/0000000000 at 192.168.0.100) Ended
>>> 2012-09-27 09:47:19.593467 [NOTICE] switch_core_session.c:1437 Close
>>> Channel sofia/internal/0000000000 at 192.168.0.100 [CS_DESTROY]
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:527 (
>>> sofia/internal/0000000000 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:530 (
>>> sofia/internal/0000000000 at 192.168.0.100) Running State Change CS_DESTROY
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/0000000000 at 192.168.0.100) State DESTROY
>>> 2012-09-27 09:47:19.593467 [DEBUG] mod_sofia.c:376
>>> sofia/internal/0000000000 at 192.168.0.100 SOFIA DESTROY
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:99
>>> sofia/internal/0000000000 at 192.168.0.100 Standard DESTROY
>>> 2012-09-27 09:47:19.593467 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/0000000000 at 192.168.0.100) State DESTROY going to sleep
>>>
>>> ============================================================================
>>>
>>>
>>>
>>>
>>> 2012/9/27 zhi sun <sunzhimailbox at gmail.com>
>>>
>>> thanks for response, the detailed debug log as below:
>>>
>>> version 2.1.3
>>>
>>> there are two sip client: 1002, 1010
>>>
>>> i can call 1002 from 1010 successfully.
>>>
>>> then i try to call 0911 from 1010, as you know the 1002 is a member of
>>> sales group, the 0911 will try to out call 1002 according the rule in
>>> default.xml
>>>
>>> =============================================
>>> 2012-09-27 09:34:07.533467 [NOTICE] switch_channel.c:951 New Channel
>>> sofia/internal/1010 at 192.168.0.100 [6dfa8876-0843-11e2-b574-8d63071eb0f5]
>>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_NEW
>>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_state_machine.c:416 (
>>> sofia/internal/1010 at 192.168.0.100) State NEW
>>> 2012-09-27 09:34:07.533467 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.553467 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>>> by acl "domains". Falling back to Digest auth.
>>> 2012-09-27 09:34:07.553467 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.553467 [DEBUG] sofia.c:1728 detaching session
>>> 6dfa8876-0843-11e2-b574-8d63071eb0f5
>>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:1820 Re-attaching to session
>>> 6dfa8876-0843-11e2-b574-8d63071eb0f5
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:8412 IP 192.168.0.10 Rejected
>>> by acl "domains". Falling back to Digest auth.
>>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1010 at 192.168.0.100 entering state [received][100]
>>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:6293 Remote SDP:
>>> v=0
>>> o=1010 3118 3118 IN IP4 192.168.0.10
>>> s=Talk
>>> c=IN IP4 192.168.0.10
>>> t=0 0
>>> m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
>>> a=rtpmap:112 speex/32000
>>> a=fmtp:112 vbr=on
>>> a=rtpmap:111 speex/16000
>>> a=fmtp:111 vbr=on
>>> a=rtpmap:110 speex/8000
>>> a=fmtp:110 vbr=on
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-11
>>>
>>> 2012-09-27 09:34:07.573471 [DEBUG] sofia.c:6506 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_NEW -> CS_INIT
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_INIT
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/1010 at 192.168.0.100) State INIT
>>> 2012-09-27 09:34:07.573471 [DEBUG] mod_sofia.c:86
>>> sofia/internal/1010 at 192.168.0.100 SOFIA INIT
>>> 2012-09-27 09:34:07.573471 [DEBUG] mod_sofia.c:126 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_INIT -> CS_ROUTING
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:437 (
>>> sofia/internal/1010 at 192.168.0.100) State INIT going to sleep
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_ROUTING
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_channel.c:1964 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change DOWN -> RINGING
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/1010 at 192.168.0.100) State ROUTING
>>> 2012-09-27 09:34:07.573471 [DEBUG] mod_sofia.c:149
>>> sofia/internal/1010 at 192.168.0.100 SOFIA ROUTING
>>> 2012-09-27 09:34:07.573471 [DEBUG] switch_core_state_machine.c:117
>>> sofia/internal/1010 at 192.168.0.100 Standard ROUTING
>>> 2012-09-27 09:34:07.573471 [INFO] mod_dialplan_xml.c:485 Processing 1010
>>> <1010>->0911 in context default
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->unloop]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS) [unloop]
>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [unloop]
>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->tod_example] continue=true
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/Time Match (PASS)
>>> [tod_example] break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action set(open=true)
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->holiday_example] continue=true
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Date/TimeMatch (FAIL)
>>> [holiday_example] break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->global-intercept] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [global-intercept] destination_number(0911) =~ /^886$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group-intercept] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group-intercept] destination_number(0911) =~ /^\*8$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->intercept-ext] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [intercept-ext] destination_number(0911) =~ /^\*\*(\d+)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->redial]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [redial]
>>> destination_number(0911) =~ /^(redial|870)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->global]
>>> continue=true
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>>> ${call_debug}(false) =~ /^true$/ break=never
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [global]
>>> ${sip_has_crypto}() =~
>>> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Absolute Condition [global]
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>>
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> hash(insert/${domain_name}-last_dial/global/${uuid})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->snom-demo-2] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-2]
>>> destination_number(0911) =~ /^9001$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->snom-demo-1] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [snom-demo-1]
>>> destination_number(0911) =~ /^9000$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->eavesdrop] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>>> destination_number(0911) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->eavesdrop] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [eavesdrop]
>>> destination_number(0911) =~ /^779$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->call_return] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [call_return]
>>> destination_number(0911) =~ /^\*69$|^869$|^lcr$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->del-group] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [del-group]
>>> destination_number(0911) =~ /^80(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->add-group] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [add-group]
>>> destination_number(0911) =~ /^81(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->call-group-simo] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [call-group-simo] destination_number(0911) =~ /^82(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->call-group-order] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [call-group-order] destination_number(0911) =~ /^83(\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->extension-intercom] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [extension-intercom] destination_number(0911) =~ /^8(10[01][0-9])$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->Local_Extension] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [Local_Extension] destination_number(0911) =~ /^(10[01][0-9])$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->Local_Extension_Skinny] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [Local_Extension_Skinny] destination_number(0911) =~ /^(11[01][0-9])$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group_dial_sales] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group_dial_sales] destination_number(0911) =~ /^2000$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group_dial_support] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group_dial_support] destination_number(0911) =~ /^2001$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->group_dial_billing] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [group_dial_billing] destination_number(0911) =~ /^2002$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->operator]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [operator]
>>> destination_number(0911) =~ /^(operator|0)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->vmain]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [vmain]
>>> destination_number(0911) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing [default->sip_uri]
>>> continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL) [sip_uri]
>>> destination_number(0911) =~ /^sip:(.*)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->nb_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [nb_conferences] destination_number(0911) =~ /^(30\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->wb_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [wb_conferences] destination_number(0911) =~ /^(31\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->uwb_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [uwb_conferences] destination_number(0911) =~ /^(32\d{2})$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->cdquality_conferences] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [cdquality_conferences] destination_number(0911) =~ /^(33\d{2})$/
>>> break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->freeswitch_public_conf_via_sip] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (FAIL)
>>> [freeswitch_public_conf_via_sip] destination_number(0911) =~
>>> /^9(888|8888|1616|3232)$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 parsing
>>> [default->mad_boss_intercom] continue=false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Regex (PASS)
>>> [mad_boss_intercom] destination_number(0911) =~ /^0911$/ break=on-false
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> set(conference_auto_outcall_caller_id_name=Mad Boss1)
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> set(conference_auto_outcall_caller_id_number=0911)
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> set(conference_auto_outcall_timeout=60)
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> set(conference_auto_outcall_flags=mute)
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> set(conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
>>> 2 a s1 transfer::intercept:${uuid} inline'})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> set(sip_exclude_contact=${network_addr})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> conference_set_auto_outcall(${group_call(sales)})
>>> Dialplan: sofia/internal/1010 at 192.168.0.100 Action
>>> conference(madboss_intercom1 at default+flags{endconf|deaf})
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:167 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_ROUTING -> CS_EXECUTE
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:446 (
>>> sofia/internal/1010 at 192.168.0.100) State ROUTING going to sleep
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_EXECUTE
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:453 (
>>> sofia/internal/1010 at 192.168.0.100) State EXECUTE
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_sofia.c:242
>>> sofia/internal/1010 at 192.168.0.100 SOFIA EXECUTE
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_state_machine.c:209
>>> sofia/internal/1010 at 192.168.0.100 Standard EXECUTE
>>> EXECUTE sofia/internal/1010 at 192.168.0.100 set(open=true)
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET [open]=[true]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-spymap/1010/6dfa8876-0843-11e2-b574-8d63071eb0f5)
>>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/1010/0911)
>>> EXECUTE sofia/internal/1010 at 192.168.0.100hash(insert/192.168.0.100-last_dial/global/6dfa8876-0843-11e2-b574-8d63071eb0f5)
>>> EXECUTE sofia/internal/1010 at 192.168.0.100 export(RFC2822_DATE=Thu, 27
>>> Sep 2012 09:34:07 +0800)
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_channel.c:1118 EXPORT
>>> (export_vars) [RFC2822_DATE]=[Thu, 27 Sep 2012 09:34:07 +0800]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_caller_id_name=Mad Boss1)
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET
>>> [conference_auto_outcall_caller_id_name]=[Mad Boss1]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_caller_id_number=0911)
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET
>>> [conference_auto_outcall_caller_id_number]=[0911]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_timeout=60)
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET
>>> [conference_auto_outcall_timeout]=[60]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_flags=mute)
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET
>>> [conference_auto_outcall_flags]=[mute]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100set(conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
>>> 2 a s1 transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline'})
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET
>>> [conference_auto_outcall_prefix]=[{sip_auto_answer=true,execute_on_answer='bind_meta_app
>>> 2 a s1 transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline'}]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100set(sip_exclude_contact=192.168.0.10)
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_dptools.c:1319
>>> sofia/internal/1010 at 192.168.0.100 SET
>>> [sip_exclude_contact]=[192.168.0.10]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:2390
>>> Application conference_set_auto_outcall Requires media! pre_answering
>>> channel sofia/internal/1010 at 192.168.0.100
>>> 2012-09-27 09:34:07.593468 [INFO] switch_core_session.c:2392 Sending
>>> early media
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:112:32000:20:0]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:111:16000:20:0]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [speex:110:8000:20:0]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[G722:9:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
>>> [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3077 Set Codec
>>> sofia/internal/1010 at 192.168.0.100 GSM/8000 20 ms 160 samples 13200 bits
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_codec.c:111
>>> sofia/internal/1010 at 192.168.0.100 Original read codec set to GSM:3
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:5219 Set 2833 dtmf
>>> send/recv payload to 101
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3327 AUDIO RTP [
>>> sofia/internal/1010 at 192.168.0.100] 192.168.0.100 port 19426 ->
>>> 192.168.0.10 port 7078 codec: 3 ms: 20
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_rtp.c:1927 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3591 Set 2833 dtmf send
>>> payload to 101
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3597 Set 2833 dtmf
>>> receive payload to 101
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia_glue.c:3624
>>> sofia/internal/1010 at 192.168.0.100 Set rtp dtmf delay to 40
>>> 2012-09-27 09:34:07.593468 [NOTICE] sofia_glue.c:4226 Pre-Answer
>>> sofia/internal/1010 at 192.168.0.100!
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_channel.c:3092 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change RINGING -> EARLY
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_sofia.c:2730 Ring SDP:
>>> v=0
>>> o=FreeSWITCH 1348690221 1348690222 IN IP4 192.168.0.100
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.0.100
>>> t=0 0
>>> m=audio 19426 RTP/AVP 3 101
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:759 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100conference_set_auto_outcall([sip_invite_domain=192.168.0.100,
>>> presence_id=1000 at 192.168.0.100]sofia/internal/sip:1000 at 192.168.0.20:5060<
>>> http://sip:1000 at 192.168.0.20:5060> ,[sip_invite_domain=192.168.0.100,
>>> presence_id=1001 at 192.168.0.100]error/user_not_registered
>>> ,[sip_invite_domain=192.168.0.100,
>>> presence_id=1002 at 192.168.0.100]error/user_not_registered
>>> ,[sip_invite_domain=192.168.0.100,
>>> presence_id=1003 at 192.168.0.100]error/user_not_registered
>>> ,[sip_invite_domain=192.168.0.100,
>>> presence_id=1004 at 192.168.0.100]error/user_not_registered)
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1010 at 192.168.0.100 entering state [early][183]
>>> EXECUTE sofia/internal/1010 at 192.168.0.100conference(madboss_intercom1 at default
>>> +flags{endconf|deaf})
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_sofia.c:836 Local SDP
>>> sofia/internal/1010 at 192.168.0.100:
>>> v=0
>>> o=FreeSWITCH 1348690221 1348690223 IN IP4 192.168.0.100
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.0.100
>>> t=0 0
>>> m=audio 19426 RTP/AVP 3 101
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:759 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_channel.c:3351 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change EARLY -> ACTIVE
>>> 2012-09-27 09:34:07.593468 [NOTICE] mod_conference.c:7211 Channel [
>>> sofia/internal/1010 at 192.168.0.100] has been answered
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.593468 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1010 at 192.168.0.100 entering state [completed][200]
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8315 using channel
>>> sound prefix:
>>> /home/sunzhi/sunzhi/freeswitch/freeswitch-1.2.3/rel/sounds/en/us/callie
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:7092 Raw Codec
>>> Activation Success L16 at 8000hz 1 channel 20ms
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:7137 Raw Codec
>>> Activation Success L16 at 8000hz 1 channel 20ms
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:1922 Setup timer
>>> success interval: 20  samples: 160
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_codec.c:219
>>> sofia/internal/1010 at 192.168.0.100 Push codec L16:70
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '0' to 'mute'
>>> 2012-09-27 09:34:07.593468 [INFO] switch_ivr_async.c:194 Digit parser
>>> mod_conference: Setting realm to 'conf'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 0/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bab8
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '*' to 'deaf mute'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding */conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bae8
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '9' to 'energy up'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 9/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bb18
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '8' to 'energy equ'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 8/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bb48
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '7' to 'energy dn'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 7/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bb78
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '3' to 'vol talk up'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 3/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bba8
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '2' to 'vol talk zero'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 2/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bbd8
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '1' to 'vol talk dn'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 1/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bc08
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '6' to 'vol listen up'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 6/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bc38
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '5' to 'vol listen zero'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 5/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bc68
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '4' to 'vol listen dn'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding 4/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bc98
>>> 2012-09-27 09:34:07.593468 [INFO] mod_conference.c:8869
>>> sofia/internal/1010 at 192.168.0.100 binding '#' to 'hangup'
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_async.c:288 Digit parser
>>> mod_conference: binding #/conf/0 callback: 0x7f7bcbde4640 data:
>>> 0x7f7b8002bcc8
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_core_session.c:759 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:3474 Setup timer
>>> soft success interval: 20  samples: 160
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>>> session specific variables
>>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1000 at 192.168.0.100]sofia
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>>> session specific variables
>>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1000 at 192.168.0.100]sofia]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1001 at 192.168.0.100]error
>>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1001 at 192.168.0.100]error]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>>> session specific variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1002 at 192.168.0.100]error
>>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1002 at 192.168.0.100]error]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>>> session specific variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] mod_conference.c:6815 Launching BG
>>> Thread for outcall
>>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2005 Parsing
>>> global variables
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1003 at 192.168.0.100]error
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1003 at 192.168.0.100]error]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [sip_auto_answer]=[true]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_event.c:1569 Parsing variable
>>> [execute_on_answer]=[bind_meta_app 2 a s1
>>> transfer::intercept:6dfa8876-0843-11e2-b574-8d63071eb0f5 inline]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:2425 Parsing
>>> session specific variables
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 09:34:07.593468 [ERR] switch_ivr_originate.c:2440 Parse Error!
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>> 2012-09-27 09:34:07.593468 [ERR] switch_core_session.c:408 Could not
>>> locate channel type presence_id=1004 at 192.168.0.100]error
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: DESTINATION_OUT_OF_ORDER
>>> 2012-09-27 09:34:07.593468 [NOTICE] switch_ivr_originate.c:2591 Cannot
>>> create outgoing channel of type [presence_id=1004 at 192.168.0.100]error]
>>> cause: [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [DEBUG] switch_ivr_originate.c:3508 Originate
>>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>>> 2012-09-27 09:34:07.593468 [ERR] mod_conference.c:6614 Cannot create
>>> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> 2012-09-27 09:34:07.613471 [DEBUG] mod_local_stream.c:417 Opening Stream
>>> [moh/8000] 8000hz
>>> 2012-09-27 09:34:07.613471 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.613471 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.613471 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:07.633467 [DEBUG] sofia.c:6282 Channel
>>> sofia/internal/1010 at 192.168.0.100 entering state [ready][200]
>>> 2012-09-27 09:34:07.693467 [DEBUG] switch_rtp.c:3596 Correct ip/port
>>> confirmed.
>>> 2012-09-27 09:34:07.713468 [DEBUG] mod_conference.c:4288 Queueing file
>>> 'tone_stream://%(500,0,640)' for play
>>> 2012-09-27 09:34:10.733467 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_channel.c:2950 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change ACTIVE -> HANGUP
>>> 2012-09-27 09:34:10.753468 [NOTICE] sofia.c:711 Hangup
>>> sofia/internal/1010 at 192.168.0.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_channel.c:2973 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [KILL]
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:10.753468 [DEBUG] mod_conference.c:3777 Channel leaving
>>> conference, cause: NORMAL_CLEARING
>>> 2012-09-27 09:34:10.753468 [DEBUG] mod_conference.c:7645
>>> sofia/internal/1010 at 192.168.0.100 skip receive message [UNBRIDGE]
>>> (channel is hungup already)
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_codec.c:244
>>> sofia/internal/1010 at 192.168.0.100 Restore previous codec GSM:3.
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:2553
>>> sofia/internal/1010 at 192.168.0.100 skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:453 (
>>> sofia/internal/1010 at 192.168.0.100) State EXECUTE going to sleep
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_HANGUP
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/1010 at 192.168.0.100) State HANGUP
>>> 2012-09-27 09:34:10.753468 [DEBUG] mod_sofia.c:483 Channel
>>> sofia/internal/1010 at 192.168.0.100 hanging up, cause: NORMAL_CLEARING
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:48
>>> sofia/internal/1010 at 192.168.0.100 Standard HANGUP, cause:
>>> NORMAL_CLEARING
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/1010 at 192.168.0.100) State HANGUP going to sleep
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:429 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_REPORTING
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/1010 at 192.168.0.100) State REPORTING
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:92
>>> sofia/internal/1010 at 192.168.0.100 Standard REPORTING, cause:
>>> NORMAL_CLEARING
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:703 (
>>> sofia/internal/1010 at 192.168.0.100) State REPORTING going to sleep
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:423 (
>>> sofia/internal/1010 at 192.168.0.100) State Change CS_REPORTING ->
>>> CS_DESTROY
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_session.c:1415 Session 3 (
>>> sofia/internal/1010 at 192.168.0.100) Locked, Waiting on external entities
>>> 2012-09-27 09:34:10.753468 [NOTICE] switch_core_session.c:1433 Session 3
>>> (sofia/internal/1010 at 192.168.0.100) Ended
>>> 2012-09-27 09:34:10.753468 [NOTICE] switch_core_session.c:1437 Close
>>> Channel sofia/internal/1010 at 192.168.0.100 [CS_DESTROY]
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:527 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change HANGUP -> DOWN
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:530 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_DESTROY
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/1010 at 192.168.0.100) State DESTROY
>>> 2012-09-27 09:34:10.753468 [DEBUG] mod_sofia.c:376
>>> sofia/internal/1010 at 192.168.0.100 SOFIA DESTROY
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:99
>>> sofia/internal/1010 at 192.168.0.100 Standard DESTROY
>>> 2012-09-27 09:34:10.753468 [DEBUG] switch_core_state_machine.c:540 (
>>> sofia/internal/1010 at 192.168.0.100) State DESTROY going to sleep
>>> 2012-09-27 09:34:10.773471 [NOTICE] mod_conference.c:2369 Ending pending
>>> outcall channels for Conference: 'madboss_intercom1'
>>> 2012-09-27 09:34:10.773471 [DEBUG] mod_conference.c:2461 Write Lock ON
>>> 2012-09-27 09:34:10.773471 [DEBUG] mod_conference.c:2464 Write Lock OFF
>>>
>>> ======================================================================
>>>
>>>
>>> 2012/9/27 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>> That is not nearly enough log file, you need to supply the entire log of
>>> the call in full debug mode
>>>
>>> On Wed, Sep 26, 2012 at 8:13 AM, zhi sun <sunzhimailbox at gmail.com>
>>> wrote:
>>>
>>> in the default diaplan, the 0911 conference call that make a out call to
>>> a group doesn't work for the latest and version 2.1.3. the out call always
>>> failed because of CHAN_NOT_IMPLEMENTED,
>>>
>>> but it works fine for version 1.06
>>>
>>> the same problem happens with the 5001 in dialplan (default.xml)
>>>
>>> it is very easy to reproduce this issue: i just get the correct version,
>>> make, make install, make cd-sounds-install.
>>>
>>> the log with problem looks like below:
>>>
>>> ===========================================
>>>
>>> 2012-09-26 15:37:10.973469 [NOTICE] switch_ivr.c:1748 Transfer
>>> sofia/internal/1000 at 192.168.0.100 to XML[1010 at default]
>>> 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/0000000000 at 192.168.0.100 [BREAK]
>>> 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2950 (
>>> sofia/internal/1010 at 192.168.0.100) Callstate Change RINGING -> HANGUP
>>> 2012-09-26 15:37:10.993469 [NOTICE] switch_ivr_originate.c:3330 Hangup
>>> sofia/internal/1010 at 192.168.0.100 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2973 Send signal
>>> sofia/internal/1010 at 192.168.0.100 [KILL]
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1010 at 192.168.0.100 [BREAK]
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_ivr_originate.c:3506 Originate
>>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_state_machine.c:398 (
>>> sofia/internal/1010 at 192.168.0.100) Running State Change CS_HANGUP
>>> 2012-09-26 15:37:10.993469 [ERR] mod_conference.c:6626 Cannot create
>>> outgoing channel, cause: ORIGINATOR_CANCEL
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2950 (
>>> sofia/internal/1000 at 192.168.0.100) Callstate Change ACTIVE -> HANGUP
>>> 2012-09-26 15:37:10.993469 [NOTICE] mod_conference.c:6629 Hangup
>>> sofia/internal/1000 at 192.168.0.100 [CS_ROUTING] [ORIGINATOR_CANCEL]
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2973 Send signal
>>> sofia/internal/1000 at 192.168.0.100 [KILL]
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_state_machine.c:638 (
>>> sofia/internal/1010 at 192.168.0.100) State HANGUP
>>> 2012-09-26 15:37:10.993469 [DEBUG] switch_core_session.c:1210 Send
>>> signal sofia/internal/1000 at 192.168.0.100 [BREAK]
>>>
>>> ============================================
>>>
>>> i am new to freeswitch, Is there anything i missed to let the version
>>> 2.1.3 as well as the latest (from git) work for the 0911 and 5001 dialplan?
>>> both of them make an out call from a conference.
>>>
>>> thanks
>>> -zhisun
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> 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
>>>
>>>
>>>
>>>
>>> --
>>> 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-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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From lconroy at insensate.co.uk  Fri Sep 28 02:11:31 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Thu, 27 Sep 2012 23:11:31 +0100
Subject: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is
	registered at another freeswitch.
In-Reply-To: <CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>
References: <5064C421.9010206@telefaks.de>
	<CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>
Message-ID: <D7C14AA8-D575-408A-BC4C-E0FD2DCD683E@insensate.co.uk>

Hi there,
 in addition, to have the VPN'd fS (box2) show up as extension 400 in the box1 fS, goto
the gateway definition (on box2) and make sure that <param name="extension" value="400"/>
and <param name="extension-in-contact" value="true"/> are set.
For extra points, look in the wiki for "auto_to_user".

I'm also curious about the results of sofia_contact */400
(probably for the same reason ;).

all the best,
  Lawrence

On 27 Sep 2012, at 22:47, Michael Collins wrote:
> Just for kicks, try this:
> 
> sofia_contact */400
> 
> I'm curious about something.
> -MC
> 
> On Thu, Sep 27, 2012 at 2:24 PM, Peter Steinbach <lists at telefaks.de> wrote:
> 
>> Hello,
>> 
>> I have 2 Freeswitch boxes.
>> Box 2 is registering as number 400 via VPN to box 1 (like a regular
>> phone). I can see in the sip_registrations database that the phone is
>> registered. And Box2 shows "
>>   vpn::sip1.mydomain.com       gateway
>> sip:400 at 10.8.0.1:5075      REGED
>> So at the first glance everything seems ok.
>> 
>> However in my vpn profile
>>   sofia status profile internalvpn
>> the Freeswitch box2 is not shown as a registration entry.
>> 
>> I think this is the reason why we receive the following when we try to
>> call the box 2 (number 400):
>> EXECUTE sofia/internal/200 at sip1.mydomain.com
>> bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
>> 2012-09-27 23:09:58.149196 [DEBUG] switch_channel.c:1062
>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
>> [dialed_ext]=[400] to event
>> 2012-09-27 23:09:58.149196 [DEBUG] switch_ivr_originate.c:1961 Parsing
>> global variables
>> 2012-09-27 23:09:58.149196 [DEBUG] switch_event.c:1478 Parsing variable
>> [sip_invite_domain]=[sip1.mydomain.com]
>> 2012-09-27 23:09:58.169252 [DEBUG] switch_core_session.c:924 Send signal
>> sofia/internal/200 at sip1.mydomain.com [BREAK]
>> 2012-09-27 23:09:58.269299 [DEBUG] switch_channel.c:1062
>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
>> [dialed_ext]=[400] to event
>> 2012-09-27 23:09:58.269299 [DEBUG] switch_ivr_originate.c:1961 Parsing
>> global variables
>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
>> [presence_id]=[400 at sip1.mydomain.com]
>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
>> [transfer_fallback_extension]=[400]
>> 2012-09-27 23:09:58.269299 [DEBUG] sofia.c:6040 Channel
>> sofia/internal/200 at sip1.mydomain.com entering state [early][180]
>> 2012-09-27 23:09:58.269299 [NOTICE] switch_ivr_originate.c:2544 Cannot
>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>> 
>> 
>> freeswitch at internal> sofia_contact 400
>> error/user_not_registered
>> freeswitch at internal> sofia_contact 200
>> sofia/internal/sip:200 at 192.168.178.105:2048
>> ;line=1vz0u23u;fs_nat=yes;fs_path=sip%3A200%40217.24.xx.xxx%3A20393%3Bline%3D1vz0u23u
>> So 200 seems ok, 400 not.
>> 
>> When I look at the contact field in the sip_registrations table, I see
>> "user"<sip:gw+sip1.mydomain.com at 10.8.0.42:5080;transport=udp;gw=
>> sip1.mydomain.com>
>> so this does not reflect the number 400.
>> 
>> Here is my gateway config from box2 which registers at box1
>> (sip1.mydomain.com)
>> # cat sip1.xml
>> <include>
>>  <gateway name="sip1.mydomain.com">
>>    <param name="from-domain" value="sip1.mydomain.com"/>
>>    <param name="username" value="400"/>
>>    <param name="from-user" value="400"/>
>>    <param name="extension" value="400"/>
>>    <param name="realm" value="sip1.mydomain.com"/>
>>    <!-- 5075 registers to the internal VPN profile -->
>>    <param name="proxy" value="10.8.0.1:5075"/>
>>    <param name="password" value="xxxxxxxx"/>
>>    <param name="expire-seconds" value="3600"/>
>>    <param name="register" value="true"/>
>>    <param name="register-transport" value="udp"/>
>>    <param name="retry-seconds" value="20"/>
>>    <param name="caller-id-in-from" value="true"/>
>>  </gateway>
>> </include>
>> 
>> 
>> My question is: How can I change the gateway definition, so that box2
>> shows up as a regular registration with number 400? Or am I missing
>> something different?
>> 
>> --
>> 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 gabe at gundy.org  Fri Sep 28 03:34:55 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 27 Sep 2012 17:34:55 -0600
Subject: [Freeswitch-users] Saving phone information from a Polycom IP501
In-Reply-To: <1342813962.5783.88.camel@mythtv>
References: <1342813962.5783.88.camel@mythtv>
Message-ID: <CAMwZ-tLrLz5hK-Yk2B1BNLiROqtyQ0msphZs89Bb7-RQRUrmdA@mail.gmail.com>

On Fri, Jul 20, 2012 at 1:52 PM, Todd Bailey <toddb at toddbailey.net> wrote:
> Just wondering if it's possible to download data from a phone and save
> it into separate files or into a database?

The Polycom documentation is pretty good, but it's extensive. I'd say
the learning curve for the Polycom platform is pretty steep, but worth
while. Anyway, just my 2 cents.

Gabe


From asaad2 at gmail.com  Fri Sep 28 04:54:47 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 27 Sep 2012 20:54:47 -0400
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CALNZuwntPp-h9VJELy++T+8wsWn4mf+3u-HSwuq+S5TwdYZynw@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
	<CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
	<CALNZuwntPp-h9VJELy++T+8wsWn4mf+3u-HSwuq+S5TwdYZynw@mail.gmail.com>
Message-ID: <CAAdTMa8rdx9cBE5TUZccN7y3KRe9+WkRfVv-COrOXdO-bz_D4g@mail.gmail.com>

If you change your web server to run on port 22. Will it still detect that
your http protocol or will show it as an ssh protocol?
On Sep 27, 2012 3:56 PM, "Avi Marcus" <avi at avimarcus.net> wrote:

> nmap offers service detection:
>
>
> # nmap -sV some-domain.com
> ...
> 22/tcp   open   ssh         OpenSSH 5.3p1 Debian 3ubuntu7 (protocol 2.0)
> 80/tcp   open   http        nginx web server 0.8.54
> ...
> 5060/tcp open   sip         (SIP end point; Status: 200 OK)
> 5080/tcp open   sip         (SIP end point; Status: 200 OK)
> ...
> Nmap done: 1 IP address (1 host up) scanned in 90.91 seconds
>
> vs 5 seconds for plain scan. But still, it exists.
>
> -Avi
>
>
>
> On Thu, Sep 27, 2012 at 9:27 PM, BookBag <asaad2 at gmail.com> wrote:
>
>> when nmap finds a port open, it looks in its database of what protocol is
>> likely to be running on that port. It doesnt actually test if the standard
>> protocol is running on that port.
>>
>>
>>
>>
>>
>> On Thu, Sep 27, 2012 at 12:11 PM, Nelson Camargo <bigx333 at gmail.com>wrote:
>>
>>> Ever heard about nmap? lol
>>> On 27 Sep 2012, at 5:52 PM, BookBag wrote:
>>>
>>> How will they know what protocol I'm running on that port?
>>> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>>
>>>> This is classic wardialing and is very common. Don't worry, your port
>>>> change won't slow down people who really want to get in ;)
>>>>
>>>>
>>>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>>>
>>>>> I had the same issue. There are hackers continuously scanning public
>>>>> ip's for known ports then trying to register devices using the default
>>>>> extensions and passwords "1234". After noticing this in my logs I just
>>>>> changed the default external sip port from 5080 to something else.
>>>>>
>>>>> Security through obscurity if you will.
>>>>> P.S. I was also using fail2ban
>>>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>>>> wrote:
>>>>>
>>>>>> Hi There,
>>>>>>  welcome to our world; hope it didn't cost too much.
>>>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>>>> them.
>>>>>> I'd guess that this particular bunch are coming from IP addresses
>>>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>>>> appeared to originate.
>>>>>>
>>>>>> 1st rule of fight club: Firewalls are no use for a server that is
>>>>>> going to listen for requests from the Internet and allow authenticated
>>>>>> calls to be placed from any IP address.
>>>>>>
>>>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up
>>>>>> and works just fine [unless you're using Windoz, in which case God hates
>>>>>> you**].
>>>>>>
>>>>>> For more refined control (if you know where your external contacts
>>>>>> are coming from) ...
>>>>>>
>>>>>> Consider setting up ACLs (nailing down the IP address ranges from
>>>>>> which you'll accept incalls) in autoload/acl.conf.xml -- the "domains"
>>>>>> definition there is one place to add in your external correspondents.
>>>>>>
>>>>>> Also, consider using cidr= parameters in your directory folder for
>>>>>> each of your users (if they will only attempt to register or place calls
>>>>>> from given address ranges).
>>>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>>>
>>>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and
>>>>>> take it from there.
>>>>>>
>>>>>> BTW, you WILL be confused by setting explicit ACLs on registration --
>>>>>> leave that one commented out until you know what it actually does, as it's
>>>>>> probably not what you expect. Several strong cups of coffee and protracted
>>>>>> meditation may help.
>>>>>>
>>>>>> Main message:
>>>>>> -- Immediately - fix the passwords so they're not easy to guess [as
>>>>>> the bad guys *will* try again and again until they get it right].
>>>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>>>> proposed. <======= MOST IMPORTANT
>>>>>> -- lose the belief that firewalls are going to help protect an
>>>>>> Internet-listening server as, logically, they can't
>>>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>>>> logfile, and wonder how you got away with it for so long.
>>>>>>
>>>>>> all the best,
>>>>>>   Lawrence
>>>>>> ** There was apparently a talk on how Windows users could get
>>>>>> something close to a fail2ban-style setup (IIRC, it was on the weekly conf
>>>>>> call a while back)
>>>>>>
>>>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado
>>>>>> wrote:
>>>>>> > I really think that people give way too much importance to
>>>>>> firewalls,
>>>>>> > specially stateless ones, blocking ports isn't going to do much for
>>>>>> you
>>>>>> > unless you are trying to hide vulnerable services behind it.
>>>>>> >
>>>>>> > They used the extension 1000 to make the calls so I would say:
>>>>>> activate
>>>>>> > log-auth-failures on your profile, setup a fail2ban and get stronger
>>>>>> > passwords.
>>>>>> >
>>>>>> > If you want to go further you can use a stateful firewall limiting
>>>>>> > connections and setup a IDS(recommend snort)
>>>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net>
>>>>>> wrote:
>>>>>> >
>>>>>> >>
>>>>>> >> Hey All,
>>>>>> >>
>>>>>> >>
>>>>>> >> I just got an email from Frontier that there were several attempts
>>>>>> to
>>>>>> >> make international calls.
>>>>>> >>
>>>>>> >>
>>>>>> >> I checked the log file and verified that somehow someone was able
>>>>>> to get
>>>>>> >> access to FS from the internet.
>>>>>> >>
>>>>>> >>
>>>>>> >> here is a sample of the log
>>>>>> >>
>>>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941 New
>>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>>> Ring-Ready
>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>>> switch_ivr_originate.c:519
>>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>>> Pre-Answer
>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>>> switch_ivr_originate.c:3303
>>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>>>> 16:30:29.916821
>>>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941 New
>>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>>> Ring-Ready
>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>>> switch_ivr_originate.c:519
>>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been answered
>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>>> Pre-Answer
>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>>> switch_ivr_originate.c:3303
>>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941 New
>>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>>>> >>
>>>>>> >>
>>>>>> >> At this point I'm at a loss how this is happening as I have
>>>>>> multiple
>>>>>> >> firewalls in place that limit port access.
>>>>>> >>
>>>>>> >> Can someone provide a few pointers on how to better secure FS
>>>>>> running on
>>>>>> >> Linux systems?
>>>>>> >>
>>>>>> >>
>>>>>> >> thanks
>>>>>> >>
>>>>>> >>
>>>>>> >> --
>>>>>> >> -
>>>>>> >> -
>>>>>> >> -    Best Regards,
>>>>>> >> -
>>>>>> >> -            Todd Bailey
>>>>>> >> -
>>>>>> >> -
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> _________________________________________________________________________
>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>> >> consulting at freeswitch.org
>>>>>> >> http://www.freeswitchsolutions.com
>>>>>> >>
>>>>>> >> 
>>>>>> >> 
>>>>>> >>
>>>>>> >> Official FreeSWITCH Sites
>>>>>> >> http://www.freeswitch.org
>>>>>> >> http://wiki.freeswitch.org
>>>>>> >> http://www.cluecon.com
>>>>>> >>
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >
>>>>>> _________________________________________________________________________
>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>> > consulting at freeswitch.org
>>>>>> > http://www.freeswitchsolutions.com
>>>>>> >
>>>>>> > 
>>>>>> > 
>>>>>> >
>>>>>> > Official FreeSWITCH Sites
>>>>>> > http://www.freeswitch.org
>>>>>> > http://wiki.freeswitch.org
>>>>>> > http://www.cluecon.com
>>>>>> >
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Fri Sep 28 05:14:05 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Sep 2012 03:14:05 +0200
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa8rdx9cBE5TUZccN7y3KRe9+WkRfVv-COrOXdO-bz_D4g@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
	<CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
	<CALNZuwntPp-h9VJELy++T+8wsWn4mf+3u-HSwuq+S5TwdYZynw@mail.gmail.com>
	<CAAdTMa8rdx9cBE5TUZccN7y3KRe9+WkRfVv-COrOXdO-bz_D4g@mail.gmail.com>
Message-ID: <CALNZuwkL6etx_9f2LgW=c5jCpHN4aegEFF6A1XsX53=SZTcxfA@mail.gmail.com>

If you do service identification, "nmap -sV", then yes. It takes a lot
longer than straight up port scanning, though.

-Avi

On Fri, Sep 28, 2012 at 2:54 AM, BookBag <asaad2 at gmail.com> wrote:

> If you change your web server to run on port 22. Will it still detect that
> your http protocol or will show it as an ssh protocol?
>  On Sep 27, 2012 3:56 PM, "Avi Marcus" <avi at avimarcus.net> wrote:
>
>> nmap offers service detection:
>>
>>
>> # nmap -sV some-domain.com
>> ...
>> 22/tcp   open   ssh         OpenSSH 5.3p1 Debian 3ubuntu7 (protocol 2.0)
>> 80/tcp   open   http        nginx web server 0.8.54
>> ...
>> 5060/tcp open   sip         (SIP end point; Status: 200 OK)
>> 5080/tcp open   sip         (SIP end point; Status: 200 OK)
>> ...
>> Nmap done: 1 IP address (1 host up) scanned in 90.91 seconds
>>
>> vs 5 seconds for plain scan. But still, it exists.
>>
>> -Avi
>>
>>
>>
>> On Thu, Sep 27, 2012 at 9:27 PM, BookBag <asaad2 at gmail.com> wrote:
>>
>>> when nmap finds a port open, it looks in its database of what protocol
>>> is likely to be running on that port. It doesnt actually test if the
>>> standard protocol is running on that port.
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Sep 27, 2012 at 12:11 PM, Nelson Camargo <bigx333 at gmail.com>wrote:
>>>
>>>> Ever heard about nmap? lol
>>>> On 27 Sep 2012, at 5:52 PM, BookBag wrote:
>>>>
>>>> How will they know what protocol I'm running on that port?
>>>> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>>>
>>>>> This is classic wardialing and is very common. Don't worry, your port
>>>>> change won't slow down people who really want to get in ;)
>>>>>
>>>>>
>>>>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>>>>
>>>>>> I had the same issue. There are hackers continuously scanning public
>>>>>> ip's for known ports then trying to register devices using the default
>>>>>> extensions and passwords "1234". After noticing this in my logs I just
>>>>>> changed the default external sip port from 5080 to something else.
>>>>>>
>>>>>> Security through obscurity if you will.
>>>>>> P.S. I was also using fail2ban
>>>>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi There,
>>>>>>>  welcome to our world; hope it didn't cost too much.
>>>>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>>>>> them.
>>>>>>> I'd guess that this particular bunch are coming from IP addresses
>>>>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>>>>> appeared to originate.
>>>>>>>
>>>>>>> 1st rule of fight club: Firewalls are no use for a server that is
>>>>>>> going to listen for requests from the Internet and allow authenticated
>>>>>>> calls to be placed from any IP address.
>>>>>>>
>>>>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up
>>>>>>> and works just fine [unless you're using Windoz, in which case God hates
>>>>>>> you**].
>>>>>>>
>>>>>>> For more refined control (if you know where your external contacts
>>>>>>> are coming from) ...
>>>>>>>
>>>>>>> Consider setting up ACLs (nailing down the IP address ranges from
>>>>>>> which you'll accept incalls) in autoload/acl.conf.xml -- the "domains"
>>>>>>> definition there is one place to add in your external correspondents.
>>>>>>>
>>>>>>> Also, consider using cidr= parameters in your directory folder for
>>>>>>> each of your users (if they will only attempt to register or place calls
>>>>>>> from given address ranges).
>>>>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>>>>
>>>>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and
>>>>>>> take it from there.
>>>>>>>
>>>>>>> BTW, you WILL be confused by setting explicit ACLs on registration
>>>>>>> -- leave that one commented out until you know what it actually does, as
>>>>>>> it's probably not what you expect. Several strong cups of coffee and
>>>>>>> protracted meditation may help.
>>>>>>>
>>>>>>> Main message:
>>>>>>> -- Immediately - fix the passwords so they're not easy to guess [as
>>>>>>> the bad guys *will* try again and again until they get it right].
>>>>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>>>>> proposed. <======= MOST IMPORTANT
>>>>>>> -- lose the belief that firewalls are going to help protect an
>>>>>>> Internet-listening server as, logically, they can't
>>>>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>>>>> logfile, and wonder how you got away with it for so long.
>>>>>>>
>>>>>>> all the best,
>>>>>>>   Lawrence
>>>>>>> ** There was apparently a talk on how Windows users could get
>>>>>>> something close to a fail2ban-style setup (IIRC, it was on the weekly conf
>>>>>>> call a while back)
>>>>>>>
>>>>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado
>>>>>>> wrote:
>>>>>>> > I really think that people give way too much importance to
>>>>>>> firewalls,
>>>>>>> > specially stateless ones, blocking ports isn't going to do much
>>>>>>> for you
>>>>>>> > unless you are trying to hide vulnerable services behind it.
>>>>>>> >
>>>>>>> > They used the extension 1000 to make the calls so I would say:
>>>>>>> activate
>>>>>>> > log-auth-failures on your profile, setup a fail2ban and get
>>>>>>> stronger
>>>>>>> > passwords.
>>>>>>> >
>>>>>>> > If you want to go further you can use a stateful firewall limiting
>>>>>>> > connections and setup a IDS(recommend snort)
>>>>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net>
>>>>>>> wrote:
>>>>>>> >
>>>>>>> >>
>>>>>>> >> Hey All,
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> I just got an email from Frontier that there were several
>>>>>>> attempts to
>>>>>>> >> make international calls.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> I checked the log file and verified that somehow someone was able
>>>>>>> to get
>>>>>>> >> access to FS from the internet.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> here is a sample of the log
>>>>>>> >>
>>>>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>>>> Ring-Ready
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>>>> switch_ivr_originate.c:519
>>>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been
>>>>>>> answered
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>>>> Pre-Answer
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>>>> switch_ivr_originate.c:3303
>>>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>>>>> 16:30:29.916821
>>>>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>>>> Ring-Ready
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>>>> switch_ivr_originate.c:519
>>>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been
>>>>>>> answered
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>>>> Pre-Answer
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>>>> switch_ivr_originate.c:3303
>>>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> At this point I'm at a loss how this is happening as I have
>>>>>>> multiple
>>>>>>> >> firewalls in place that limit port access.
>>>>>>> >>
>>>>>>> >> Can someone provide a few pointers on how to better secure FS
>>>>>>> running on
>>>>>>> >> Linux systems?
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> thanks
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> --
>>>>>>> >> -
>>>>>>> >> -
>>>>>>> >> -    Best Regards,
>>>>>>> >> -
>>>>>>> >> -            Todd Bailey
>>>>>>> >> -
>>>>>>> >> -
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> _________________________________________________________________________
>>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>>> >> consulting at freeswitch.org
>>>>>>> >> http://www.freeswitchsolutions.com
>>>>>>> >>
>>>>>>> >> 
>>>>>>> >> 
>>>>>>> >>
>>>>>>> >> Official FreeSWITCH Sites
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >> http://wiki.freeswitch.org
>>>>>>> >> http://www.cluecon.com
>>>>>>> >>
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >
>>>>>>> _________________________________________________________________________
>>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>>> > consulting at freeswitch.org
>>>>>>> > http://www.freeswitchsolutions.com
>>>>>>> >
>>>>>>> > 
>>>>>>> > 
>>>>>>> >
>>>>>>> > Official FreeSWITCH Sites
>>>>>>> > http://www.freeswitch.org
>>>>>>> > http://wiki.freeswitch.org
>>>>>>> > http://www.cluecon.com
>>>>>>> >
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 28 05:52:38 2012
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 28 Sep 2012 09:52:38 +0800
Subject: [Freeswitch-users] Empty wiki and my head
In-Reply-To: <CAKzWOxUPqjbns7Bo1Zb-2kD_Bb9hK6x=15s0V_As00zZOMO_Ag@mail.gmail.com>
References: <CAF=swh_LmBERMqTx4goR4aoGrKfk4ZzMTAh-CoYmGWS90N7Uog@mail.gmail.com>
	<CAKzWOxUPqjbns7Bo1Zb-2kD_Bb9hK6x=15s0V_As00zZOMO_Ag@mail.gmail.com>
Message-ID: <60FA3CD2B49B498A89E981AA84C863CD@gmail.com>

Looks like mod_sonar is just like a real Sonar that - 

- You setup a remote server which can echo
- Generate some tones on the local server and call the remote server
- When the echo back use some tone_detect or VAD things to help check possible network problems

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Friday, September 28, 2012 at 12:25 AM, Michael Collins wrote:

> No documentation currently exists, therefore we need community assistance. I don't believe mod_html5 is complete, so don't sweat that one. I've never heard of mod_sonar. I know the Moc wrote mod_abstraction and that he talked about it on a conference call quite a while back but I have no further information. You might want to track him down (IRC: Moc) and see if he can give you more info. I do know that mod_xml_scgi is like mod_xml_curl but using SCGI instead. Again, I've never had a chance to use it, so if anyone has please step up and help with the documentation. 
> 
> Feel free to experiment, look at the source code, etc. and put some basic information on the wiki. I'm sure the dev team would be happy to answer specific questions. As a show of good faith it might be nice to create the necessary wiki pages as stubs and then add what you know. Then let the collective community answer specific questions and fill in the blanks.
> 
> -MC
> 
> 
> 
> On Tue, Sep 25, 2012 at 7:15 PM, Valery Kalinin <valery.kalinin at gmail.com (mailto:valery.kalinin at gmail.com)> wrote:
> > I cannot find any documentation for these modules:
> > mod_abstraction
> > mod_sonar
> > mod_html5
> > mod_xml_scgi
> > What they do and how to use them?
> > 
> > _________________________________________________________________________
> > 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
> 
> 


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From ntomer at newgen.co.in  Fri Sep 28 10:28:44 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 28 Sep 2012 11:58:44 +0530
Subject: [Freeswitch-users] Help needed in a Contact Center solution
In-Reply-To: <CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>
References: <011601cd9cac$2919d840$7b4d88c0$@co.in>	<CAMwZ-tJnaHnwq1P+Z91O+_vz54xmPjdYu9BtJ3vm0cbPt5KNZQ@mail.gmail.com>
	<CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>
Message-ID: <013301cd9d42$82ebdb70$88c39250$@co.in>

Hi Michael,

 

Thanks for your response. I have been able to configure IVR and it works
great. But I am not able to understand how to ask caller to enter some
identification number, followed by hash. And then enter that value in
database.

 

Any pointers?

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, September 27, 2012 10:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help needed in a Contact Center solution

 

Don't forget about our two books:
http://link.packtpub.com/nuIOlX
https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tele
phony-systems/book

They discuss items 1 and 2 in pretty good detail, especially chapter 6 of
the "bridge" book (it has a bridge on the cover)
-MC

On Thu, Sep 27, 2012 at 5:53 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

On Thu, Sep 27, 2012 at 6:32 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> 1. An end-customer calls, the call is handled by mod_ivr.
> 2. Customer is presented with a menu, she makes selection and also enters
> some identification. Her call is parked on a pre-decided extension and an
> entry is made in a database for this call. This entry will hold the
> selections made by the caller and the extension where call is parked.

<SNIP>


> Please guide me on how can do points 1 & 2.

1) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr#General_Concept
2) http://wiki.freeswitch.org/wiki/Mod_db#mod_db ?


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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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 avi at avimarcus.net  Fri Sep 28 10:39:27 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Sep 2012 08:39:27 +0200
Subject: [Freeswitch-users] Empty wiki and my head
In-Reply-To: <60FA3CD2B49B498A89E981AA84C863CD@gmail.com>
References: <CAF=swh_LmBERMqTx4goR4aoGrKfk4ZzMTAh-CoYmGWS90N7Uog@mail.gmail.com>
	<CAKzWOxUPqjbns7Bo1Zb-2kD_Bb9hK6x=15s0V_As00zZOMO_Ag@mail.gmail.com>
	<60FA3CD2B49B498A89E981AA84C863CD@gmail.com>
Message-ID: <CALNZuwm2j_qe=c+jF7NrnrqyRdcK4vFMMwmFbam59oS=9iVRkQ@mail.gmail.com>

Can we get some sample code for using mod_sonar? I've been wanting to test
the quality of DIDs, and/or term carriers, for a while now.
-Avi


On Fri, Sep 28, 2012 at 3:52 AM, Seven Du <dujinfang at gmail.com> wrote:

>  Looks like mod_sonar is just like a real Sonar that -
>
> - You setup a remote server which can echo
> - Generate some tones on the local server and call the remote server
> - When the echo back use some tone_detect or VAD things to help check
> possible network problems
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Friday, September 28, 2012 at 12:25 AM, Michael Collins wrote:
>
> No documentation currently exists, therefore we need community assistance.
> I don't believe mod_html5 is complete, so don't sweat that one. I've never
> heard of mod_sonar. I know the Moc wrote mod_abstraction and that he talked
> about it on a conference call quite a while back but I have no further
> information. You might want to track him down (IRC: Moc) and see if he can
> give you more info. I do know that mod_xml_scgi is like mod_xml_curl but
> using SCGI instead. Again, I've never had a chance to use it, so if anyone
> has please step up and help with the documentation.
>
> Feel free to experiment, look at the source code, etc. and put some basic
> information on the wiki. I'm sure the dev team would be happy to answer
> specific questions. As a show of good faith it might be nice to create the
> necessary wiki pages as stubs and then add what you know. Then let the
> collective community answer specific questions and fill in the blanks.
>
> -MC
>
>
>
> On Tue, Sep 25, 2012 at 7:15 PM, Valery Kalinin <valery.kalinin at gmail.com>wrote:
>
> I cannot find any documentation for these modules:
> mod_abstraction
> mod_sonar
> mod_html5
> mod_xml_scgi
> What they do and how to use them?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
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From lists at telefaks.de  Fri Sep 28 12:02:33 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 28 Sep 2012 10:02:33 +0200
Subject: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is
 registered at another freeswitch.
In-Reply-To: <CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>
References: <5064C421.9010206@telefaks.de>
	<CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>
Message-ID: <50655999.9070302@telefaks.de>

Here it is:

freeswitch at internal> sofia_contact */400
sofia/internalvpn/sip:gw+sip1.mydomain.com
@10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com

What can we read from this?


On 09/27/12 23:47, Michael Collins wrote:
> Just for kicks, try this:
>
> sofia_contact */400
>
> I'm curious about something.
> -MC
>
> On Thu, Sep 27, 2012 at 2:24 PM, Peter Steinbach <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello,
>
>     I have 2 Freeswitch boxes.
>     Box 2 is registering as number 400 via VPN to box 1 (like a regular
>     phone). I can see in the sip_registrations database that the phone is
>     registered. And Box2 shows "
>        vpn::sip1.mydomain.com <http://sip1.mydomain.com>       gateway
>     sip:400 at 10.8.0.1:5075 <http://sip:400 at 10.8.0.1:5075>      REGED
>     So at the first glance everything seems ok.
>
>     However in my vpn profile
>        sofia status profile internalvpn
>     the Freeswitch box2 is not shown as a registration entry.
>
>     I think this is the reason why we receive the following when we try to
>     call the box 2 (number 400):
>     EXECUTE sofia/internal/200 at sip1.mydomain.com
>     <mailto:200 at sip1.mydomain.com>
>     bridge({sip_invite_domain=sip1.mydomain.com
>     <http://sip1.mydomain.com>}user/400 at sip1.mydomain.com
>     <mailto:400 at sip1.mydomain.com>)
>     2012-09-27 23:09:58.149196 [DEBUG] switch_channel.c:1062
>     sofia/internal/200 at sip1.mydomain.com
>     <mailto:200 at sip1.mydomain.com> EXPORTING[export_vars]
>     [dialed_ext]=[400] to event
>     2012-09-27 23:09:58.149196 [DEBUG] switch_ivr_originate.c:1961 Parsing
>     global variables
>     2012-09-27 23:09:58.149196 [DEBUG] switch_event.c:1478 Parsing
>     variable
>     [sip_invite_domain]=[sip1.mydomain.com <http://sip1.mydomain.com>]
>     2012-09-27 23:09:58.169252 [DEBUG] switch_core_session.c:924 Send
>     signal
>     sofia/internal/200 at sip1.mydomain.com
>     <mailto:200 at sip1.mydomain.com> [BREAK]
>     2012-09-27 23:09:58.269299 [DEBUG] switch_channel.c:1062
>     sofia/internal/200 at sip1.mydomain.com
>     <mailto:200 at sip1.mydomain.com> EXPORTING[export_vars]
>     [dialed_ext]=[400] to event
>     2012-09-27 23:09:58.269299 [DEBUG] switch_ivr_originate.c:1961 Parsing
>     global variables
>     2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing
>     variable
>     [presence_id]=[400 at sip1.mydomain.com <mailto:400 at sip1.mydomain.com>]
>     2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing
>     variable
>     [transfer_fallback_extension]=[400]
>     2012-09-27 23:09:58.269299 [DEBUG] sofia.c:6040 Channel
>     sofia/internal/200 at sip1.mydomain.com
>     <mailto:200 at sip1.mydomain.com> entering state [early][180]
>     2012-09-27 23:09:58.269299 [NOTICE] switch_ivr_originate.c:2544 Cannot
>     create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>
>
>     freeswitch at internal> sofia_contact 400
>     error/user_not_registered
>     freeswitch at internal> sofia_contact 200
>     sofia/internal/sip:200 at 192.168.178.105:2048;line=1vz0u23u;fs_nat=yes;fs_path=sip%3A200%40217.24.xx.xxx%3A20393%3Bline%3D1vz0u23u
>     So 200 seems ok, 400 not.
>
>     When I look at the contact field in the sip_registrations table, I see
>     "user"<sip:gw+sip1.mydomain.com at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com
>     <http://sip1.mydomain.com>>
>     so this does not reflect the number 400.
>
>     Here is my gateway config from box2 which registers at box1
>     (sip1.mydomain.com <http://sip1.mydomain.com>)
>     # cat sip1.xml
>     <include>
>       <gateway name="sip1.mydomain.com <http://sip1.mydomain.com>">
>         <param name="from-domain" value="sip1.mydomain.com
>     <http://sip1.mydomain.com>"/>
>         <param name="username" value="400"/>
>         <param name="from-user" value="400"/>
>         <param name="extension" value="400"/>
>         <param name="realm" value="sip1.mydomain.com
>     <http://sip1.mydomain.com>"/>
>         <!-- 5075 registers to the internal VPN profile -->
>         <param name="proxy" value="10.8.0.1:5075 <http://10.8.0.1:5075>"/>
>         <param name="password" value="xxxxxxxx"/>
>         <param name="expire-seconds" value="3600"/>
>         <param name="register" value="true"/>
>         <param name="register-transport" value="udp"/>
>         <param name="retry-seconds" value="20"/>
>         <param name="caller-id-in-from" value="true"/>
>       </gateway>
>     </include>
>
>
>     My question is: How can I change the gateway definition, so that box2
>     shows up as a regular registration with number 400? Or am I missing
>     something different?
>
>     --
>     With kind regards
>     Peter Steinbach
>
>     Telefaks Services GmbH
>     mailto:lists <mailto:lists> (att) telefaks.de <http://telefaks.de>
>     Internet: www.telefaks.de <http://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


-- 
With kind regards
Peter Steinbach 

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


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From lists at telefaks.de  Fri Sep 28 12:17:24 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 28 Sep 2012 10:17:24 +0200
Subject: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is
 registered at another freeswitch.
In-Reply-To: <D7C14AA8-D575-408A-BC4C-E0FD2DCD683E@insensate.co.uk>
References: <5064C421.9010206@telefaks.de>
	<CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>
	<D7C14AA8-D575-408A-BC4C-E0FD2DCD683E@insensate.co.uk>
Message-ID: <50655D14.6040704@telefaks.de>

Hello Lawrence,

I set the

  <param name="extension-in-contact" value="true"/>

to the gateway definition and now the registration string in
sip_registarations looks much better:
   "user"<sip:400 at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com>

However with sofia_contact I still have the same problem:
  freeswitch at internal> sofia_contact 400
  error/user_not_registered
  freeswitch at internal> sofia_contact */400
 
sofia/internalvpn/sip:400 at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com

But still the same when dialling:
EXECUTE sofia/internal/200 at sip1.mydomain.com
bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
2012-09-28 10:07:29.089083 [DEBUG] switch_channel.c:1062
sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
[dialed_ext]=[400] to event
2012-09-28 10:07:29.089083 [DEBUG] switch_ivr_originate.c:1961 Parsing
global variables
2012-09-28 10:07:29.089083 [DEBUG] switch_event.c:1478 Parsing variable
[sip_invite_domain]=[sip1.mydomain.com]
2012-09-28 10:07:29.189083 [DEBUG] switch_channel.c:1062
sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
[dialed_ext]=[400] to event
2012-09-28 10:07:29.189083 [DEBUG] switch_ivr_originate.c:1961 Parsing
global variables
2012-09-28 10:07:29.189083 [DEBUG] switch_event.c:1478 Parsing variable
[presence_id]=[400 at sip1.mydomain.com]
2012-09-28 10:07:29.189083 [DEBUG] switch_event.c:1478 Parsing variable
[transfer_fallback_extension]=[400]
2012-09-28 10:07:29.189083 [NOTICE] switch_ivr_originate.c:2544 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]

Best regards
Peter

On 09/28/12 00:11, Lawrence Conroy wrote:
> Hi there,
>  in addition, to have the VPN'd fS (box2) show up as extension 400 in the box1 fS, goto
> the gateway definition (on box2) and make sure that <param name="extension" value="400"/>
> and <param name="extension-in-contact" value="true"/> are set.
> For extra points, look in the wiki for "auto_to_user".
>
> I'm also curious about the results of sofia_contact */400
> (probably for the same reason ;).
>
> all the best,
>   Lawrence
>
> On 27 Sep 2012, at 22:47, Michael Collins wrote:
>> Just for kicks, try this:
>>
>> sofia_contact */400
>>
>> I'm curious about something.
>> -MC
>>
>> On Thu, Sep 27, 2012 at 2:24 PM, Peter Steinbach <lists at telefaks.de> wrote:
>>
>>> Hello,
>>>
>>> I have 2 Freeswitch boxes.
>>> Box 2 is registering as number 400 via VPN to box 1 (like a regular
>>> phone). I can see in the sip_registrations database that the phone is
>>> registered. And Box2 shows "
>>>   vpn::sip1.mydomain.com       gateway
>>> sip:400 at 10.8.0.1:5075      REGED
>>> So at the first glance everything seems ok.
>>>
>>> However in my vpn profile
>>>   sofia status profile internalvpn
>>> the Freeswitch box2 is not shown as a registration entry.
>>>
>>> I think this is the reason why we receive the following when we try to
>>> call the box 2 (number 400):
>>> EXECUTE sofia/internal/200 at sip1.mydomain.com
>>> bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_channel.c:1062
>>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
>>> [dialed_ext]=[400] to event
>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_ivr_originate.c:1961 Parsing
>>> global variables
>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_event.c:1478 Parsing variable
>>> [sip_invite_domain]=[sip1.mydomain.com]
>>> 2012-09-27 23:09:58.169252 [DEBUG] switch_core_session.c:924 Send signal
>>> sofia/internal/200 at sip1.mydomain.com [BREAK]
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_channel.c:1062
>>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
>>> [dialed_ext]=[400] to event
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_ivr_originate.c:1961 Parsing
>>> global variables
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
>>> [presence_id]=[400 at sip1.mydomain.com]
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
>>> [transfer_fallback_extension]=[400]
>>> 2012-09-27 23:09:58.269299 [DEBUG] sofia.c:6040 Channel
>>> sofia/internal/200 at sip1.mydomain.com entering state [early][180]
>>> 2012-09-27 23:09:58.269299 [NOTICE] switch_ivr_originate.c:2544 Cannot
>>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>>>
>>>
>>> freeswitch at internal> sofia_contact 400
>>> error/user_not_registered
>>> freeswitch at internal> sofia_contact 200
>>> sofia/internal/sip:200 at 192.168.178.105:2048
>>> ;line=1vz0u23u;fs_nat=yes;fs_path=sip%3A200%40217.24.xx.xxx%3A20393%3Bline%3D1vz0u23u
>>> So 200 seems ok, 400 not.
>>>
>>> When I look at the contact field in the sip_registrations table, I see
>>> "user"<sip:gw+sip1.mydomain.com at 10.8.0.42:5080;transport=udp;gw=
>>> sip1.mydomain.com>
>>> so this does not reflect the number 400.
>>>
>>> Here is my gateway config from box2 which registers at box1
>>> (sip1.mydomain.com)
>>> # cat sip1.xml
>>> <include>
>>>  <gateway name="sip1.mydomain.com">
>>>    <param name="from-domain" value="sip1.mydomain.com"/>
>>>    <param name="username" value="400"/>
>>>    <param name="from-user" value="400"/>
>>>    <param name="extension" value="400"/>
>>>    <param name="realm" value="sip1.mydomain.com"/>
>>>    <!-- 5075 registers to the internal VPN profile -->
>>>    <param name="proxy" value="10.8.0.1:5075"/>
>>>    <param name="password" value="xxxxxxxx"/>
>>>    <param name="expire-seconds" value="3600"/>
>>>    <param name="register" value="true"/>
>>>    <param name="register-transport" value="udp"/>
>>>    <param name="retry-seconds" value="20"/>
>>>    <param name="caller-id-in-from" value="true"/>
>>>  </gateway>
>>> </include>
>>>
>>>
>>> My question is: How can I change the gateway definition, so that box2
>>> shows up as a regular registration with number 400? Or am I missing
>>> something different?
>>>
>>> --
>>> 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
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From miha at softnet.si  Fri Sep 28 12:32:51 2012
From: miha at softnet.si (Miha)
Date: Fri, 28 Sep 2012 10:32:51 +0200
Subject: [Freeswitch-users] xml_curl
Message-ID: <506560B3.2080006@softnet.si>

Hi,

I am using xml_curl for dialplan. For programing language I am using PHP.

How can I access variables like Caller-Destination-Number, Channel-Name, 
variables with -, because in php I can not use - for variables. I can 
only access variables which are defined like variable_sip_full_from etc.


Thanks for help!

Miha


From nathandownes at hotmail.com  Fri Sep 28 12:35:41 2012
From: nathandownes at hotmail.com (Mr Nathan Downes)
Date: Fri, 28 Sep 2012 18:35:41 +1000
Subject: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is
	registered at another freeswitch.
In-Reply-To: <50655D14.6040704@telefaks.de>
References: <5064C421.9010206@telefaks.de>	<CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>	<D7C14AA8-D575-408A-BC4C-E0FD2DCD683E@insensate.co.uk>
	<50655D14.6040704@telefaks.de>
Message-ID: <BLU170-DS1460C4DF0BB490985A79ADD7820@phx.gbl>

Hi Peter,

Thing next step is to edit your equivalent to 

/usr/local/freeswitch/conf/directory/default.xml

Then change

<param name="dial-string"
value="{sip_invite_domain=${domain_name},presence_id=${dialed_user}@${dialed
_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

To

<param name="dial-string"
value="{sip_invite_domain=${domain_name},presence_id=${dialed_user}@${dialed
_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>

Appears some recent change needs this to search through multiple profiles
even though you match user/domain

P.S. I could be wrong!

Thanks,

Nathan

-----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, 28 September 2012 6:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is
registered at another freeswitch.

Hello Lawrence,

I set the

  <param name="extension-in-contact" value="true"/>

to the gateway definition and now the registration string in
sip_registarations looks much better:
   "user"<sip:400 at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com>

However with sofia_contact I still have the same problem:
  freeswitch at internal> sofia_contact 400
  error/user_not_registered
  freeswitch at internal> sofia_contact */400
 
sofia/internalvpn/sip:400 at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com

But still the same when dialling:
EXECUTE sofia/internal/200 at sip1.mydomain.com
bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
2012-09-28 10:07:29.089083 [DEBUG] switch_channel.c:1062
sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
[dialed_ext]=[400] to event
2012-09-28 10:07:29.089083 [DEBUG] switch_ivr_originate.c:1961 Parsing
global variables
2012-09-28 10:07:29.089083 [DEBUG] switch_event.c:1478 Parsing variable
[sip_invite_domain]=[sip1.mydomain.com]
2012-09-28 10:07:29.189083 [DEBUG] switch_channel.c:1062
sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
[dialed_ext]=[400] to event
2012-09-28 10:07:29.189083 [DEBUG] switch_ivr_originate.c:1961 Parsing
global variables
2012-09-28 10:07:29.189083 [DEBUG] switch_event.c:1478 Parsing variable
[presence_id]=[400 at sip1.mydomain.com]
2012-09-28 10:07:29.189083 [DEBUG] switch_event.c:1478 Parsing variable
[transfer_fallback_extension]=[400]
2012-09-28 10:07:29.189083 [NOTICE] switch_ivr_originate.c:2544 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]

Best regards
Peter

On 09/28/12 00:11, Lawrence Conroy wrote:
> Hi there,
>  in addition, to have the VPN'd fS (box2) show up as extension 400 in 
> the box1 fS, goto the gateway definition (on box2) and make sure that 
> <param name="extension" value="400"/> and <param
name="extension-in-contact" value="true"/> are set.
> For extra points, look in the wiki for "auto_to_user".
>
> I'm also curious about the results of sofia_contact */400 (probably 
> for the same reason ;).
>
> all the best,
>   Lawrence
>
> On 27 Sep 2012, at 22:47, Michael Collins wrote:
>> Just for kicks, try this:
>>
>> sofia_contact */400
>>
>> I'm curious about something.
>> -MC
>>
>> On Thu, Sep 27, 2012 at 2:24 PM, Peter Steinbach <lists at telefaks.de>
wrote:
>>
>>> Hello,
>>>
>>> I have 2 Freeswitch boxes.
>>> Box 2 is registering as number 400 via VPN to box 1 (like a regular 
>>> phone). I can see in the sip_registrations database that the phone 
>>> is registered. And Box2 shows "
>>>   vpn::sip1.mydomain.com       gateway
>>> sip:400 at 10.8.0.1:5075      REGED
>>> So at the first glance everything seems ok.
>>>
>>> However in my vpn profile
>>>   sofia status profile internalvpn
>>> the Freeswitch box2 is not shown as a registration entry.
>>>
>>> I think this is the reason why we receive the following when we try 
>>> to call the box 2 (number 400):
>>> EXECUTE sofia/internal/200 at sip1.mydomain.com
>>> bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.c
>>> om)
>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_channel.c:1062 
>>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars] 
>>> [dialed_ext]=[400] to event
>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_ivr_originate.c:1961 
>>> Parsing global variables
>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_event.c:1478 Parsing 
>>> variable [sip_invite_domain]=[sip1.mydomain.com]
>>> 2012-09-27 23:09:58.169252 [DEBUG] switch_core_session.c:924 Send 
>>> signal sofia/internal/200 at sip1.mydomain.com [BREAK]
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_channel.c:1062 
>>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars] 
>>> [dialed_ext]=[400] to event
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_ivr_originate.c:1961 
>>> Parsing global variables
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing 
>>> variable [presence_id]=[400 at sip1.mydomain.com]
>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing 
>>> variable [transfer_fallback_extension]=[400]
>>> 2012-09-27 23:09:58.269299 [DEBUG] sofia.c:6040 Channel 
>>> sofia/internal/200 at sip1.mydomain.com entering state [early][180]
>>> 2012-09-27 23:09:58.269299 [NOTICE] switch_ivr_originate.c:2544 
>>> Cannot create outgoing channel of type [error] cause: 
>>> [USER_NOT_REGISTERED]
>>>
>>>
>>> freeswitch at internal> sofia_contact 400 error/user_not_registered 
>>> freeswitch at internal> sofia_contact 200
>>> sofia/internal/sip:200 at 192.168.178.105:2048
>>> ;line=1vz0u23u;fs_nat=yes;fs_path=sip%3A200%40217.24.xx.xxx%3A20393%
>>> 3Bline%3D1vz0u23u
>>> So 200 seems ok, 400 not.
>>>
>>> When I look at the contact field in the sip_registrations table, I 
>>> see "user"<sip:gw+sip1.mydomain.com at 10.8.0.42:5080;transport=udp;gw=
>>> sip1.mydomain.com>
>>> so this does not reflect the number 400.
>>>
>>> Here is my gateway config from box2 which registers at box1
>>> (sip1.mydomain.com)
>>> # cat sip1.xml
>>> <include>
>>>  <gateway name="sip1.mydomain.com">
>>>    <param name="from-domain" value="sip1.mydomain.com"/>
>>>    <param name="username" value="400"/>
>>>    <param name="from-user" value="400"/>
>>>    <param name="extension" value="400"/>
>>>    <param name="realm" value="sip1.mydomain.com"/>
>>>    <!-- 5075 registers to the internal VPN profile -->
>>>    <param name="proxy" value="10.8.0.1:5075"/>
>>>    <param name="password" value="xxxxxxxx"/>
>>>    <param name="expire-seconds" value="3600"/>
>>>    <param name="register" value="true"/>
>>>    <param name="register-transport" value="udp"/>
>>>    <param name="retry-seconds" value="20"/>
>>>    <param name="caller-id-in-from" value="true"/>  </gateway> 
>>> </include>
>>>
>>>
>>> My question is: How can I change the gateway definition, so that 
>>> box2 shows up as a regular registration with number 400? Or am I 
>>> missing something different?
>>>
>>> --
>>> 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-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
>


--
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 avi at avimarcus.net  Fri Sep 28 12:45:05 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Sep 2012 10:45:05 +0200
Subject: [Freeswitch-users] xml_curl
In-Reply-To: <506560B3.2080006@softnet.si>
References: <506560B3.2080006@softnet.si>
Message-ID: <CALNZuw=Yh6PaOu5U8P5Wc5CjE-et6K-VbaJdTgVv2eYjNiSH4w@mail.gmail.com>

In PHP they are referenced as $_POST['Caller-Caller-ID-Number']
or $_POST['variable_outside_call']... it's not a direct variable, so you
can put any kind of name inside the quotes.

-Avi


On Fri, Sep 28, 2012 at 10:32 AM, Miha <miha at softnet.si> wrote:

> Hi,
>
> I am using xml_curl for dialplan. For programing language I am using PHP.
>
> How can I access variables like Caller-Destination-Number, Channel-Name,
> variables with -, because in php I can not use - for variables. I can
> only access variables which are defined like variable_sip_full_from etc.
>
>
> Thanks for help!
>
> 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
>
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From ntomer at newgen.co.in  Fri Sep 28 13:08:22 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 28 Sep 2012 14:38:22 +0530
Subject: [Freeswitch-users] Help needed in a Contact Center solution
In-Reply-To: <CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>
References: <011601cd9cac$2919d840$7b4d88c0$@co.in>	<CAMwZ-tJnaHnwq1P+Z91O+_vz54xmPjdYu9BtJ3vm0cbPt5KNZQ@mail.gmail.com>
	<CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>
Message-ID: <016201cd9d58$d0463f80$70d2be80$@co.in>

Hi,

 

I am planning to keep a definite number of extensions (say 50) reserved for
parking calls. Is there a way, using which I can get the available parking
slots and park the call there? And then enter the parked extension number
and user selection in a database?

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, September 27, 2012 10:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help needed in a Contact Center solution

 

Don't forget about our two books:
http://link.packtpub.com/nuIOlX
https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tele
phony-systems/book

They discuss items 1 and 2 in pretty good detail, especially chapter 6 of
the "bridge" book (it has a bridge on the cover)
-MC

On Thu, Sep 27, 2012 at 5:53 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

On Thu, Sep 27, 2012 at 6:32 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> 1. An end-customer calls, the call is handled by mod_ivr.
> 2. Customer is presented with a menu, she makes selection and also enters
> some identification. Her call is parked on a pre-decided extension and an
> entry is made in a database for this call. This entry will hold the
> selections made by the caller and the extension where call is parked.

<SNIP>


> Please guide me on how can do points 1 & 2.

1) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr#General_Concept
2) http://wiki.freeswitch.org/wiki/Mod_db#mod_db ?


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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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 gabe at gundy.org  Fri Sep 28 13:09:10 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 28 Sep 2012 03:09:10 -0600
Subject: [Freeswitch-users] xml_curl
In-Reply-To: <506560B3.2080006@softnet.si>
References: <506560B3.2080006@softnet.si>
Message-ID: <CAMwZ-t+87HeQpJdrQ=eHSZ54EmX7G06ANqcej99cjm1EsWdkmg@mail.gmail.com>

On Fri, Sep 28, 2012 at 2:32 AM, Miha <miha at softnet.si> wrote:
> I am using xml_curl for dialplan. For programing language I am using PHP.
>
> How can I access variables like Caller-Destination-Number, Channel-Name,
> variables with -, because in php I can not use - for variables. I can
> only access variables which are defined like variable_sip_full_from etc.


http://wiki.freeswitch.org/wiki/Mod_xml_curl#Request

Here are the details of the POST.


Gabe


From lconroy at insensate.co.uk  Fri Sep 28 13:37:35 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Fri, 28 Sep 2012 10:37:35 +0100
Subject: [Freeswitch-users] USER_NOT_REGISTERED when freeswtch is
	registered at another freeswitch.
In-Reply-To: <50655D14.6040704@telefaks.de>
References: <5064C421.9010206@telefaks.de>
	<CAKzWOxV9xr+N2=Yxf5cZvTvjcDuUYSX9sptUc_9NWq1yA79b1A@mail.gmail.com>
	<D7C14AA8-D575-408A-BC4C-E0FD2DCD683E@insensate.co.uk>
	<50655D14.6040704@telefaks.de>
Message-ID: <2DB11020-D6CF-40D6-BD64-F4C72D53477E@insensate.co.uk>

Hi Peter, folks,
 OK -- next step;
I believe one reason for this situation is that your dialplan (on box1) tries to find the user 400 at sip1.mydomain.com,
but you're registering as 400@<IP-address>
Box1's internal sip profile does allow this (else you would see a "you need to add a user for 400 in domain 10.8.0.1"
report on the console when box 2 tried to register).

I'd also guess that, if you type "sofia status profile internal" on the console, you would see that box2
*is* shown as registered, as 400 at 10.8.0.1

When you try to call out to user 400 at sip1.mydomain.com, it can't be found in the database, hence the error.

So ... is the above what you see?

As proposed, you need to enable a single domain and DB for the internal profile,
and change the dial-string in the directory. That should nail down the dial-string, and so allow 400 to be found.

all the best,
  Lawrence

On 28 Sep 2012, at 09:17, Peter Steinbach wrote:
> Hello Lawrence,
> 
> I set the
> 
>  <param name="extension-in-contact" value="true"/>
> 
> to the gateway definition and now the registration string in
> sip_registarations looks much better:
>   "user"<sip:400 at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com>
> 
> However with sofia_contact I still have the same problem:
>  freeswitch at internal> sofia_contact 400
>  error/user_not_registered
>  freeswitch at internal> sofia_contact */400
> 
> sofia/internalvpn/sip:400 at 10.8.0.42:5080;transport=udp;gw=sip1.mydomain.com
> 
> But still the same when dialling:
> EXECUTE sofia/internal/200 at sip1.mydomain.com
> bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
> 2012-09-28 10:07:29.089083 [DEBUG] switch_channel.c:1062
> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
> [dialed_ext]=[400] to event
> 2012-09-28 10:07:29.089083 [DEBUG] switch_ivr_originate.c:1961 Parsing
> global variables
> 2012-09-28 10:07:29.089083 [DEBUG] switch_event.c:1478 Parsing variable
> [sip_invite_domain]=[sip1.mydomain.com]
> 2012-09-28 10:07:29.189083 [DEBUG] switch_channel.c:1062
> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
> [dialed_ext]=[400] to event
> 2012-09-28 10:07:29.189083 [DEBUG] switch_ivr_originate.c:1961 Parsing
> global variables
> 2012-09-28 10:07:29.189083 [DEBUG] switch_event.c:1478 Parsing variable
> [presence_id]=[400 at sip1.mydomain.com]
> 2012-09-28 10:07:29.189083 [DEBUG] switch_event.c:1478 Parsing variable
> [transfer_fallback_extension]=[400]
> 2012-09-28 10:07:29.189083 [NOTICE] switch_ivr_originate.c:2544 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 
> Best regards
> Peter
> 
> On 09/28/12 00:11, Lawrence Conroy wrote:
>> Hi there,
>> in addition, to have the VPN'd fS (box2) show up as extension 400 in the box1 fS, goto
>> the gateway definition (on box2) and make sure that <param name="extension" value="400"/>
>> and <param name="extension-in-contact" value="true"/> are set.
>> For extra points, look in the wiki for "auto_to_user".
>> 
>> I'm also curious about the results of sofia_contact */400
>> (probably for the same reason ;).
>> 
>> all the best,
>>  Lawrence
>> 
>> On 27 Sep 2012, at 22:47, Michael Collins wrote:
>>> Just for kicks, try this:
>>> 
>>> sofia_contact */400
>>> 
>>> I'm curious about something.
>>> -MC
>>> 
>>> On Thu, Sep 27, 2012 at 2:24 PM, Peter Steinbach <lists at telefaks.de> wrote:
>>> 
>>>> Hello,
>>>> 
>>>> I have 2 Freeswitch boxes.
>>>> Box 2 is registering as number 400 via VPN to box 1 (like a regular
>>>> phone). I can see in the sip_registrations database that the phone is
>>>> registered. And Box2 shows "
>>>>  vpn::sip1.mydomain.com       gateway
>>>> sip:400 at 10.8.0.1:5075      REGED
>>>> So at the first glance everything seems ok.
>>>> 
>>>> However in my vpn profile
>>>>  sofia status profile internalvpn
>>>> the Freeswitch box2 is not shown as a registration entry.
>>>> 
>>>> I think this is the reason why we receive the following when we try to
>>>> call the box 2 (number 400):
>>>> EXECUTE sofia/internal/200 at sip1.mydomain.com
>>>> bridge({sip_invite_domain=sip1.mydomain.com}user/400 at sip1.mydomain.com)
>>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_channel.c:1062
>>>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
>>>> [dialed_ext]=[400] to event
>>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_ivr_originate.c:1961 Parsing
>>>> global variables
>>>> 2012-09-27 23:09:58.149196 [DEBUG] switch_event.c:1478 Parsing variable
>>>> [sip_invite_domain]=[sip1.mydomain.com]
>>>> 2012-09-27 23:09:58.169252 [DEBUG] switch_core_session.c:924 Send signal
>>>> sofia/internal/200 at sip1.mydomain.com [BREAK]
>>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_channel.c:1062
>>>> sofia/internal/200 at sip1.mydomain.com EXPORTING[export_vars]
>>>> [dialed_ext]=[400] to event
>>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_ivr_originate.c:1961 Parsing
>>>> global variables
>>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
>>>> [presence_id]=[400 at sip1.mydomain.com]
>>>> 2012-09-27 23:09:58.269299 [DEBUG] switch_event.c:1478 Parsing variable
>>>> [transfer_fallback_extension]=[400]
>>>> 2012-09-27 23:09:58.269299 [DEBUG] sofia.c:6040 Channel
>>>> sofia/internal/200 at sip1.mydomain.com entering state [early][180]
>>>> 2012-09-27 23:09:58.269299 [NOTICE] switch_ivr_originate.c:2544 Cannot
>>>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>>>> 
>>>> 
>>>> freeswitch at internal> sofia_contact 400
>>>> error/user_not_registered
>>>> freeswitch at internal> sofia_contact 200
>>>> sofia/internal/sip:200 at 192.168.178.105:2048
>>>> ;line=1vz0u23u;fs_nat=yes;fs_path=sip%3A200%40217.24.xx.xxx%3A20393%3Bline%3D1vz0u23u
>>>> So 200 seems ok, 400 not.
>>>> 
>>>> When I look at the contact field in the sip_registrations table, I see
>>>> "user"<sip:gw+sip1.mydomain.com at 10.8.0.42:5080;transport=udp;gw=
>>>> sip1.mydomain.com>
>>>> so this does not reflect the number 400.
>>>> 
>>>> Here is my gateway config from box2 which registers at box1
>>>> (sip1.mydomain.com)
>>>> # cat sip1.xml
>>>> <include>
>>>> <gateway name="sip1.mydomain.com">
>>>>   <param name="from-domain" value="sip1.mydomain.com"/>
>>>>   <param name="username" value="400"/>
>>>>   <param name="from-user" value="400"/>
>>>>   <param name="extension" value="400"/>
>>>>   <param name="realm" value="sip1.mydomain.com"/>
>>>>   <!-- 5075 registers to the internal VPN profile -->
>>>>   <param name="proxy" value="10.8.0.1:5075"/>
>>>>   <param name="password" value="xxxxxxxx"/>
>>>>   <param name="expire-seconds" value="3600"/>
>>>>   <param name="register" value="true"/>
>>>>   <param name="register-transport" value="udp"/>
>>>>   <param name="retry-seconds" value="20"/>
>>>>   <param name="caller-id-in-from" value="true"/>
>>>> </gateway>
>>>> </include>
>>>> 
>>>> 
>>>> My question is: How can I change the gateway definition, so that box2
>>>> shows up as a regular registration with number 400? Or am I missing
>>>> something different?
>>>> 
>>>> --
>>>> 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
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
> 
> 



From gabe at gundy.org  Fri Sep 28 13:43:10 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 28 Sep 2012 03:43:10 -0600
Subject: [Freeswitch-users] Lenny's killing it.
Message-ID: <CAMwZ-tLdPfuoO4GLAm=Aw8PKQR8gEzLvZqqc0tr+kzVRrN4haQ@mail.gmail.com>

No way! Lenny is on a roll! It's too perfect (listen in order)!

1) http://itslenny.com/recording.php?file=4d8cde0cc5a4d86ccf221aa5b2903e95
2) http://itslenny.com/recording.php?file=4fdd3f445fff8e1b73456af602367b6e

Classic!


Gabe


From miha at softnet.si  Fri Sep 28 14:07:07 2012
From: miha at softnet.si (Miha)
Date: Fri, 28 Sep 2012 12:07:07 +0200
Subject: [Freeswitch-users] xml_curl
In-Reply-To: <CALNZuw=Yh6PaOu5U8P5Wc5CjE-et6K-VbaJdTgVv2eYjNiSH4w@mail.gmail.com>
References: <506560B3.2080006@softnet.si>
	<CALNZuw=Yh6PaOu5U8P5Wc5CjE-et6K-VbaJdTgVv2eYjNiSH4w@mail.gmail.com>
Message-ID: <506576CB.30805@softnet.si>

Thanks Avi.

Regards,
Miha

On 9/28/2012 10:45 AM, Avi Marcus wrote:
> In PHP they are referenced as $_POST['Caller-Caller-ID-Number'] 
> or $_POST['variable_outside_call']... it's not a direct variable, so 
> you can put any kind of name inside the quotes.
>
> -Avi
>
>
> On Fri, Sep 28, 2012 at 10:32 AM, Miha <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     Hi,
>
>     I am using xml_curl for dialplan. For programing language I am
>     using PHP.
>
>     How can I access variables like Caller-Destination-Number,
>     Channel-Name,
>     variables with -, because in php I can not use - for variables. I can
>     only access variables which are defined like
>     variable_sip_full_from etc.
>
>
>     Thanks for help!
>
>     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
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Sep 28 15:32:25 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Fri, 28 Sep 2012 15:32:25 +0400
Subject: [Freeswitch-users] GSMOPEN sms
Message-ID: <CAA2gtHBFLa-nh44cU+xtQGk87FkkTmsbbwxEh1pCNtv2eUD4mw@mail.gmail.com>

Hi,

I can't see incoming sms in event plain message

I'm using huawei e1550, Is there anything else needed apart from default
config?
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From ben at langfeld.co.uk  Fri Sep 28 15:59:16 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 28 Sep 2012 08:59:16 -0300
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAdTMa8rdx9cBE5TUZccN7y3KRe9+WkRfVv-COrOXdO-bz_D4g@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
	<CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
	<CALNZuwntPp-h9VJELy++T+8wsWn4mf+3u-HSwuq+S5TwdYZynw@mail.gmail.com>
	<CAAdTMa8rdx9cBE5TUZccN7y3KRe9+WkRfVv-COrOXdO-bz_D4g@mail.gmail.com>
Message-ID: <CAAyX+KHuh6_R4M88wLgNkLu2qnGe+aDu6HBdpXDZvS6j+yhh-A@mail.gmail.com>

I can just talk HTTP to you on port 22 and if I get back something that
looks like HTTP, I'm golden.

Regards,
Ben Langfeld


On 27 September 2012 21:54, BookBag <asaad2 at gmail.com> wrote:

> If you change your web server to run on port 22. Will it still detect that
> your http protocol or will show it as an ssh protocol?
> On Sep 27, 2012 3:56 PM, "Avi Marcus" <avi at avimarcus.net> wrote:
>
>> nmap offers service detection:
>>
>>
>> # nmap -sV some-domain.com
>> ...
>> 22/tcp   open   ssh         OpenSSH 5.3p1 Debian 3ubuntu7 (protocol 2.0)
>> 80/tcp   open   http        nginx web server 0.8.54
>> ...
>> 5060/tcp open   sip         (SIP end point; Status: 200 OK)
>> 5080/tcp open   sip         (SIP end point; Status: 200 OK)
>> ...
>> Nmap done: 1 IP address (1 host up) scanned in 90.91 seconds
>>
>> vs 5 seconds for plain scan. But still, it exists.
>>
>> -Avi
>>
>>
>>
>> On Thu, Sep 27, 2012 at 9:27 PM, BookBag <asaad2 at gmail.com> wrote:
>>
>>> when nmap finds a port open, it looks in its database of what protocol
>>> is likely to be running on that port. It doesnt actually test if the
>>> standard protocol is running on that port.
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Sep 27, 2012 at 12:11 PM, Nelson Camargo <bigx333 at gmail.com>wrote:
>>>
>>>> Ever heard about nmap? lol
>>>> On 27 Sep 2012, at 5:52 PM, BookBag wrote:
>>>>
>>>> How will they know what protocol I'm running on that port?
>>>> On Sep 27, 2012 11:42 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>>>
>>>>> This is classic wardialing and is very common. Don't worry, your port
>>>>> change won't slow down people who really want to get in ;)
>>>>>
>>>>>
>>>>> On 27 September 2012 11:55, BookBag <asaad2 at gmail.com> wrote:
>>>>>
>>>>>> I had the same issue. There are hackers continuously scanning public
>>>>>> ip's for known ports then trying to register devices using the default
>>>>>> extensions and passwords "1234". After noticing this in my logs I just
>>>>>> changed the default external sip port from 5080 to something else.
>>>>>>
>>>>>> Security through obscurity if you will.
>>>>>> P.S. I was also using fail2ban
>>>>>> On Sep 26, 2012 7:11 PM, "Lawrence Conroy" <lconroy at insensate.co.uk>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi There,
>>>>>>>  welcome to our world; hope it didn't cost too much.
>>>>>>> Frontier were pro-active, which is very good. Don't forget to thank
>>>>>>> them.
>>>>>>> I'd guess that this particular bunch are coming from IP addresses
>>>>>>> provided in the West bank and/or Gaza; that's from where my "visitors"
>>>>>>> appeared to originate.
>>>>>>>
>>>>>>> 1st rule of fight club: Firewalls are no use for a server that is
>>>>>>> going to listen for requests from the Internet and allow authenticated
>>>>>>> calls to be placed from any IP address.
>>>>>>>
>>>>>>> You MUST have reasonable passwords, plus fail2ban is easy to set up
>>>>>>> and works just fine [unless you're using Windoz, in which case God hates
>>>>>>> you**].
>>>>>>>
>>>>>>> For more refined control (if you know where your external contacts
>>>>>>> are coming from) ...
>>>>>>>
>>>>>>> Consider setting up ACLs (nailing down the IP address ranges from
>>>>>>> which you'll accept incalls) in autoload/acl.conf.xml -- the "domains"
>>>>>>> definition there is one place to add in your external correspondents.
>>>>>>>
>>>>>>> Also, consider using cidr= parameters in your directory folder for
>>>>>>> each of your users (if they will only attempt to register or place calls
>>>>>>> from given address ranges).
>>>>>>> Then enable ACLs for incalls in your sip profile(s).
>>>>>>>
>>>>>>> This is all covered on wiki.freeswitch.org -- search for ACLs and
>>>>>>> take it from there.
>>>>>>>
>>>>>>> BTW, you WILL be confused by setting explicit ACLs on registration
>>>>>>> -- leave that one commented out until you know what it actually does, as
>>>>>>> it's probably not what you expect. Several strong cups of coffee and
>>>>>>> protracted meditation may help.
>>>>>>>
>>>>>>> Main message:
>>>>>>> -- Immediately - fix the passwords so they're not easy to guess [as
>>>>>>> the bad guys *will* try again and again until they get it right].
>>>>>>> -- set up fail2ban (which has its own page on the wiki) exactly as
>>>>>>> proposed. <======= MOST IMPORTANT
>>>>>>> -- lose the belief that firewalls are going to help protect an
>>>>>>> Internet-listening server as, logically, they can't
>>>>>>> Finally, be amazed at the occasional "block" reports in the fail2ban
>>>>>>> logfile, and wonder how you got away with it for so long.
>>>>>>>
>>>>>>> all the best,
>>>>>>>   Lawrence
>>>>>>> ** There was apparently a talk on how Windows users could get
>>>>>>> something close to a fail2ban-style setup (IIRC, it was on the weekly conf
>>>>>>> call a while back)
>>>>>>>
>>>>>>> On 26 Sep 2012, at 19:54, Nelson Luiz Ferraz de Camargo Penteado
>>>>>>> wrote:
>>>>>>> > I really think that people give way too much importance to
>>>>>>> firewalls,
>>>>>>> > specially stateless ones, blocking ports isn't going to do much
>>>>>>> for you
>>>>>>> > unless you are trying to hide vulnerable services behind it.
>>>>>>> >
>>>>>>> > They used the extension 1000 to make the calls so I would say:
>>>>>>> activate
>>>>>>> > log-auth-failures on your profile, setup a fail2ban and get
>>>>>>> stronger
>>>>>>> > passwords.
>>>>>>> >
>>>>>>> > If you want to go further you can use a stateful firewall limiting
>>>>>>> > connections and setup a IDS(recommend snort)
>>>>>>> > On Sep 26, 2012 8:29 PM, "Todd Bailey" <toddb at toddbailey.net>
>>>>>>> wrote:
>>>>>>> >
>>>>>>> >>
>>>>>>> >> Hey All,
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> I just got an email from Frontier that there were several
>>>>>>> attempts to
>>>>>>> >> make international calls.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> I checked the log file and verified that somehow someone was able
>>>>>>> to get
>>>>>>> >> access to FS from the internet.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> here is a sample of the log
>>>>>>> >>
>>>>>>> >> [m [36m2012-09-23 16:30:29.916821 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>>>> >> [af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>>>> Ring-Ready
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>>>> switch_ivr_originate.c:519
>>>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been
>>>>>>> answered
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>>>> Pre-Answer
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>>>> switch_ivr_originate.c:3303
>>>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>>>> >> [m [36m2012-09-23 16:30:52.356865 [N [m [36m2012-09-23
>>>>>>> 16:30:29.916821
>>>>>>> >> [NOTICE] switch_channel.c:941 New Channel
>>>>>>> >> sofia/internal/1000 at 50.47.85.167[af778857-0188-4ed2-a82a-94ae749a02cb]
>>>>>>> >> [m [32m2012-09-23 16:30:29.916821 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168521352 in context default
>>>>>>> >> [m [36m2012-09-23 16:30:29.936831 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/01137168521352 at 192.168.1.5:5061
>>>>>>> >> [d1243a78-c464-45fa-9215-e7b85e1221fc]
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] sofia.c:6132 Ring-Ready
>>>>>>> >> sofia/internal/01137168521352 at 192.168.1.5:5061!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE] mod_sofia.c:2572
>>>>>>> Ring-Ready
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:29.956842 [NOTICE]
>>>>>>> switch_ivr_originate.c:519
>>>>>>> >> Ring Ready sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.936826 [NOTICE] sofia.c:6777 Channel
>>>>>>> >> [sofia/internal/01137168521352 at 192.168.1.5:5061] has been
>>>>>>> answered
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE] sofia_glue.c:4176
>>>>>>> Pre-Answer
>>>>>>> >> sofia/internal/1000 at 50.47.85.167!
>>>>>>> >> [m [36m2012-09-23 16:30:32.956825 [NOTICE]
>>>>>>> switch_ivr_originate.c:3303
>>>>>>> >> Channel [sofia/internal/1000 at 50.47.85.167] has been answered
>>>>>>> >> [m [36m2012-09-23 16:30:52.356865 [NOTICE] switch_channel.c:941
>>>>>>> New
>>>>>>> >> Channel sofia/internal/1000 at 50.47.85.167
>>>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168905352 in context defaultOTICE]
>>>>>>> >> switch_channel.c:941 New Channel sofia/internal/1000 at 50.47.85.167
>>>>>>> >> [4576bc76-144a-4f6f-8915-871b511c374d]
>>>>>>> >> [m [32m2012-09-23 16:30:52.376830 [INFO] mod_dialplan_xml.c:485
>>>>>>> >> Processing 1000 <1000>->01137168905352 in context default
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> At this point I'm at a loss how this is happening as I have
>>>>>>> multiple
>>>>>>> >> firewalls in place that limit port access.
>>>>>>> >>
>>>>>>> >> Can someone provide a few pointers on how to better secure FS
>>>>>>> running on
>>>>>>> >> Linux systems?
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> thanks
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> --
>>>>>>> >> -
>>>>>>> >> -
>>>>>>> >> -    Best Regards,
>>>>>>> >> -
>>>>>>> >> -            Todd Bailey
>>>>>>> >> -
>>>>>>> >> -
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> _________________________________________________________________________
>>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>>> >> consulting at freeswitch.org
>>>>>>> >> http://www.freeswitchsolutions.com
>>>>>>> >>
>>>>>>> >> 
>>>>>>> >> 
>>>>>>> >>
>>>>>>> >> Official FreeSWITCH Sites
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >> http://wiki.freeswitch.org
>>>>>>> >> http://www.cluecon.com
>>>>>>> >>
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >
>>>>>>> _________________________________________________________________________
>>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>>> > consulting at freeswitch.org
>>>>>>> > http://www.freeswitchsolutions.com
>>>>>>> >
>>>>>>> > 
>>>>>>> > 
>>>>>>> >
>>>>>>> > Official FreeSWITCH Sites
>>>>>>> > http://www.freeswitch.org
>>>>>>> > http://wiki.freeswitch.org
>>>>>>> > http://www.cluecon.com
>>>>>>> >
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From darcy at Vex.Net  Fri Sep 28 17:42:11 2012
From: darcy at Vex.Net (D'Arcy Cain)
Date: Fri, 28 Sep 2012 09:42:11 -0400
Subject: [Freeswitch-users] Hacking FS issue
In-Reply-To: <CAAyX+KHuh6_R4M88wLgNkLu2qnGe+aDu6HBdpXDZvS6j+yhh-A@mail.gmail.com>
References: <1348684084.7087.16.camel@mythtv.toddbailey.net>
	<CAPt4n68y1+A_iZfwBxAOwrrmOrRdiAQQ46a_wk1+VJUF-yoNqg@mail.gmail.com>
	<A5CBCF44-7565-43CB-8188-459D37E38B6C@insensate.co.uk>
	<CAAdTMa_-8YgvJ9oAghxXWOwNR0GUOgNp0Jj45ctbdHbebo1LRg@mail.gmail.com>
	<CAAyX+KGePayQjmKkM=ytowDTr1Qb9b2V-+ewbAga1TjtkVWnhA@mail.gmail.com>
	<CAAdTMa9m-D75on66NAtCt0KQyw36_ehxRoOOkOs-nENWXYGeYw@mail.gmail.com>
	<F53D7335-6BFF-4C2E-AED6-97AD330FB701@gmail.com>
	<CAAdTMa9P0-Y3W0T0S4epbNj5Y-XxkQS29ry1=T=xbCf3XskXaw@mail.gmail.com>
	<CALNZuwntPp-h9VJELy++T+8wsWn4mf+3u-HSwuq+S5TwdYZynw@mail.gmail.com>
	<CAAdTMa8rdx9cBE5TUZccN7y3KRe9+WkRfVv-COrOXdO-bz_D4g@mail.gmail.com>
	<CAAyX+KHuh6_R4M88wLgNkLu2qnGe+aDu6HBdpXDZvS6j+yhh-A@mail.gmail.com>
Message-ID: <20120928094211.dcdea5960cdff35cde962046@Vex.Net>

On Fri, 28 Sep 2012 08:59:16 -0300
Ben Langfeld <ben at langfeld.co.uk> wrote:
> I can just talk HTTP to you on port 22 and if I get back something that
> looks like HTTP, I'm golden.

That's right.  There is nothing magical about port 22 or 80 or any
other well known port.  Well known is all they are.  Any service can be
run on any port.  Running web servers on port 80, for example, just
save the user from typing "http://www.Vex.Net:80/ to access a page.
The web browser assumes ":80" if you don't put it there.

It works the other way too.  Many people put ssh on port 80 for when
they are out somewhere that blocks everything but web ports.  The ssh
client assumes port 22 but you can use -p to use any port you want.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From ssoni at lifesize.com  Fri Sep 28 18:51:01 2012
From: ssoni at lifesize.com (sanjay)
Date: Fri, 28 Sep 2012 07:51:01 -0700 (PDT)
Subject: [Freeswitch-users] Help on freeswitch : gtalk-sip conversion
In-Reply-To: <F3C02C5E8650B74FA0C199FA1A01D26E16482CD4BB@ausmsex00.austin.kmvtechnologies.com>
References: <F3C02C5E8650B74FA0C199FA1A01D26E16482CD4BB@ausmsex00.austin.kmvtechnologies.com>
Message-ID: <1348843861289-7583280.post@n2.nabble.com>

No reply on this but I moved with my debugging. Posting what i Found.... 
 
I found out that STUN (ICE) Binding requests coming from gtalk client were
using RFC3489 but FS response was using RFC5389. In effect the response from
FS were not quoting correct Transaction ID (First four bytes are MAGIC
COOKIE) and hence gtalk not able to associate them. I corrected this in
switch_stun.c and now I see once STUN response reach at gtalk client, it
accepts them. I do not see call getting ended in 30 seconds and no more
continuous ICE candidate Jabber messages from gtalk client!

But user experience level I am still at same place - gtalk client still not
sending its audio and RTP packets coming to it are not playing ! I think now
its just media level issue. But what ? 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-conversion-tp7582744p7583280.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Sep 28 19:47:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Sep 2012 08:47:44 -0700
Subject: [Freeswitch-users] Empty wiki and my head
In-Reply-To: <60FA3CD2B49B498A89E981AA84C863CD@gmail.com>
References: <CAF=swh_LmBERMqTx4goR4aoGrKfk4ZzMTAh-CoYmGWS90N7Uog@mail.gmail.com>
	<CAKzWOxUPqjbns7Bo1Zb-2kD_Bb9hK6x=15s0V_As00zZOMO_Ag@mail.gmail.com>
	<60FA3CD2B49B498A89E981AA84C863CD@gmail.com>
Message-ID: <CAKzWOxX44qmsAvUw3kBRsD07qht2eu_8JXgus2iQct8K7w+p4Q@mail.gmail.com>

Cool. I threw this on the wiki so that we can at least have *something*
http://wiki.freeswitch.org/wiki/Mod_sonar

-MC


On Thu, Sep 27, 2012 at 6:52 PM, Seven Du <dujinfang at gmail.com> wrote:

>  Looks like mod_sonar is just like a real Sonar that -
>
> - You setup a remote server which can echo
> - Generate some tones on the local server and call the remote server
> - When the echo back use some tone_detect or VAD things to help check
> possible network problems
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Friday, September 28, 2012 at 12:25 AM, Michael Collins wrote:
>
> No documentation currently exists, therefore we need community assistance.
> I don't believe mod_html5 is complete, so don't sweat that one. I've never
> heard of mod_sonar. I know the Moc wrote mod_abstraction and that he talked
> about it on a conference call quite a while back but I have no further
> information. You might want to track him down (IRC: Moc) and see if he can
> give you more info. I do know that mod_xml_scgi is like mod_xml_curl but
> using SCGI instead. Again, I've never had a chance to use it, so if anyone
> has please step up and help with the documentation.
>
> Feel free to experiment, look at the source code, etc. and put some basic
> information on the wiki. I'm sure the dev team would be happy to answer
> specific questions. As a show of good faith it might be nice to create the
> necessary wiki pages as stubs and then add what you know. Then let the
> collective community answer specific questions and fill in the blanks.
>
> -MC
>
>
>
> On Tue, Sep 25, 2012 at 7:15 PM, Valery Kalinin <valery.kalinin at gmail.com>wrote:
>
> I cannot find any documentation for these modules:
> mod_abstraction
> mod_sonar
> mod_html5
> mod_xml_scgi
> What they do and how to use them?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Fri Sep 28 19:55:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Sep 2012 08:55:55 -0700
Subject: [Freeswitch-users] Help needed in a Contact Center solution
In-Reply-To: <013301cd9d42$82ebdb70$88c39250$@co.in>
References: <011601cd9cac$2919d840$7b4d88c0$@co.in>
	<CAMwZ-tJnaHnwq1P+Z91O+_vz54xmPjdYu9BtJ3vm0cbPt5KNZQ@mail.gmail.com>
	<CAKzWOxUSuAQ-UT9yrnLBhEYp1tPCF+GkFaL-40mvSr4afRU8Gw@mail.gmail.com>
	<013301cd9d42$82ebdb70$88c39250$@co.in>
Message-ID: <CAKzWOxWV1TLKsgt7idiWseNguL8VsCTyQEwu9QtuBReyrec=hg@mail.gmail.com>

The XML IVR is for gathering information for call routing. However, if you
want to capture information from the user then you probably want this:

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

If you have need to put the response into a database then I recommend you
look at dialplan scripting with Lua (or another language you're familiar
with). Personally I find that Lua is pretty easy to learn. Check out
chapter 7 of the bridge book for some nice information on getting started
with Lua. I believe the wiki also has some good information about using Lua
with a backend database.

If you use the valet_park dialplan application then you can use the
valet_info API to pull information:
freeswitch at internal> valet_info
<lots>
  <lot name="valet_parking_lot">
    <extension uuid="75ca90bc-0985-11e2-8cc2-fdcd4f39ecad">6001</extension>
  </lot>
</lots>

Again, that's the kind of thing that isn't so easy doing right from the
dialplan but if you have a script then it's pretty easy once you learn how
to parse the XML and extract the information you care about.

Happy hacking!

-MC

On Thu, Sep 27, 2012 at 11:28 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

>  Hi Michael,****
>
> ** **
>
> Thanks for your response. I have been able to configure IVR and it works
> great. But I am not able to understand how to ask caller to enter some
> identification number, followed by hash. And then enter that value in
> database.****
>
> ** **
>
> Any pointers?****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, September 27, 2012 10:22 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Help needed in a Contact Center solution
> ****
>
> ** **
>
> Don't forget about our two books:
> http://link.packtpub.com/nuIOlX
>
> https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book
>
> They discuss items 1 and 2 in pretty good detail, especially chapter 6 of
> the "bridge" book (it has a bridge on the cover)
> -MC****
>
> On Thu, Sep 27, 2012 at 5:53 AM, Gabriel Gunderson <gabe at gundy.org> wrote:
> ****
>
> On Thu, Sep 27, 2012 at 6:32 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> > 1. An end-customer calls, the call is handled by mod_ivr.
> > 2. Customer is presented with a menu, she makes selection and also enters
> > some identification. Her call is parked on a pre-decided extension and an
> > entry is made in a database for this call. This entry will hold the
> > selections made by the caller and the extension where call is parked.***
> *
>
> <SNIP>****
>
>
> > Please guide me on how can do points 1 & 2.****
>
> 1)
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr#General_Concept
> 2) http://wiki.freeswitch.org/wiki/Mod_db#mod_db ?
>
>
> 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.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
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> _________________________________________________________________________
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>
> 
> 
>
> Official FreeSWITCH Sites
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> http://www.cluecon.com
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-- 
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 Sep 28 20:30:06 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 28 Sep 2012 11:30:06 -0500
Subject: [Freeswitch-users] Just a community Reminder...
Message-ID: <CC8B3ABE.544A4%krice@freeswitch.org>

Hey Guys,

Don?t forget to join us today on the FreeSWITCH conference bridge starting
at 4PM Eastern for the Friday Free For All...

Presenter: NO ONE
Topic: Whatever
Reason: Because We Can

Pop on the bridge, come and go as you please and hang out with other
FreeSWITCH users and some of the developers too...

Access the bridge just like the Weekly Wed Conf call.

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From tahir at ictinnovations.com  Sat Sep 29 05:18:23 2012
From: tahir at ictinnovations.com (Tahir Almas)
Date: Sat, 29 Sep 2012 06:18:23 +0500
Subject: [Freeswitch-users] Help on freeswitch : gtalk-sip conversion
In-Reply-To: <1348843861289-7583280.post@n2.nabble.com>
References: <F3C02C5E8650B74FA0C199FA1A01D26E16482CD4BB@ausmsex00.austin.kmvtechnologies.com>
	<1348843861289-7583280.post@n2.nabble.com>
Message-ID: <CAOL0t+Kt=Vv65wJ7=fAEa5=ddFkyRgFsAMVDn7eKGeF_DPJt-g@mail.gmail.com>

is selinux on freeswith node enabled ? it need to be disable



On 9/28/12, sanjay <ssoni at lifesize.com> wrote:
> No reply on this but I moved with my debugging. Posting what i Found....
>
> I found out that STUN (ICE) Binding requests coming from gtalk client were
> using RFC3489 but FS response was using RFC5389. In effect the response
> from
> FS were not quoting correct Transaction ID (First four bytes are MAGIC
> COOKIE) and hence gtalk not able to associate them. I corrected this in
> switch_stun.c and now I see once STUN response reach at gtalk client, it
> accepts them. I do not see call getting ended in 30 seconds and no more
> continuous ICE candidate Jabber messages from gtalk client!
>
> But user experience level I am still at same place - gtalk client still not
> sending its audio and RTP packets coming to it are not playing ! I think
> now
> its just media level issue. But what ?
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-conversion-tp7582744p7583280.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
>


-- 
*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
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From william at xofap.com  Sat Sep 29 09:12:48 2012
From: william at xofap.com (William Alianto)
Date: Sat, 29 Sep 2012 12:12:48 +0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 75, Issue 223
In-Reply-To: <mailman.21844.1348847788.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.21844.1348847788.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <50668350.2020901@xofap.com>

Hi,

I just installed 2 spans of OpenVox's A800E analog card to my server.

Here is my freetdm.conf :

[general]
global_parameter => value

[span zt FXO]
fxo-channel => 1:1
fxo-channel => 1:2
fxo-channel => 1:3
fxo-channel => 1:4
fxo-channel => 1:5
fxo-channel => 1:6
fxo-channel => 1:7
fxo-channel => 1:8
fxo-channel => 2:1
fxo-channel => 2:2
fxo-channel => 2:3
fxo-channel => 2:4

[span zt FXS]
fxs-channel => 2:5
fxs-channel => 2:6
fxs-channel => 2:7
fxs-channel => 2:8

and my freetdm.conf.xml

<configuration name="freetdm.conf" description="FreeTDM Configuration">
   <settings>
     <param name="debug" value="0"/>
     <param name="hold-music" value="$${moh_uri}"/>
   </settings>
   <!-- one entry here per openzap span -->
   <analog_spans>
     <span name="FXS">
       <!-- This is where the calls coming on this span will be directed 
-->
       <param name="dialplan" value="XML"/>
       <param name="context" value="default"/>
       <param name="enable_callerid" value="true"/> <!-- This is the 
default value anyway -->
     </span>
     <span name="FXO">
       <param name="dialplan" value="XML"/>
       <param name="context" value="default"/>
       <!-- Set this to the correct value for your area, to get the
            correct tones on the phones connected to the FXS ports -->
       <param name="tonegroup" value="us"/>
       <param name="enable_callerid" value="true"/>
     </span>
   </analog_spans>
</configuration>

Also here is my dialplan configuration :


<extension name="inbound" >
    <condition field="context" expression="public" />
    <condition field="destination_number" expression="^(\d+)$" >
        <action application="transfer" data="101 XML default" />
    </condition>
</extension>

<extension name="freetdm.9d" >
    <condition field="destination_number" expression="^9(\d+)$" >
        <action application="set" 
data="sip_h_X-accountcode=${accountcode}" />
        <action application="set" data="sip_h_X-Tag=" />
        <action application="set" data="call_direction=outbound" />
        <action application="set" data="hangup_after_bridge=true" />
        <action application="set" 
data="effective_caller_id_name=${outbound_caller_id_name}" />
        <action application="set" 
data="effective_caller_id_number=${outbound_caller_id_number}" />
        <action application="set" data="inherit_codec=true" />
        <action application="bridge" data="freetdm/1/a/$1" />
    </condition>
</extension>


But i still cannot make outbound and inbound call using the card. Can 
anyone help me?

Regards


From brian at freeswitch.org  Sat Sep 29 22:28:41 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Sep 2012 13:28:41 -0500
Subject: [Freeswitch-users] TEST
Message-ID: <8445172F-65DE-4E2F-A346-DEF2BBD18E58@freeswitch.org>

TEST
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543







From lists at kavun.ch  Sat Sep 29 22:29:26 2012
From: lists at kavun.ch (Emrah)
Date: Sat, 29 Sep 2012 14:29:26 -0400
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <EDBB9235-D08F-4FEC-A577-1D516FE6AA8F@kavun.ch>
References: <CC8A0C72.54439%krice@freeswitch.org>
	<EDBB9235-D08F-4FEC-A577-1D516FE6AA8F@kavun.ch>
Message-ID: <06F123E2-F649-4859-9675-26DBEE6FE68F@kavun.ch>

Hi all,

I am now trying to force inband DTMF on my PSTN peers alone. I tried the following with no luck:
<action application="start_dtmf" />
<action application="export" data="dtmf_type=inband" />

The delay is still there and I get the following output in my console for a single DTMF:

2012-09-29 19:22:57.357913 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 2:1040
2012-09-29 19:22:57.357913 [DEBUG] switch_ivr_bridge.c:393 Send signal sofia/external/1234567890 [BREAK]
2012-09-29 19:22:57.377912 [DEBUG] switch_rtp.c:2736 Send start packet for [2] ts=748000 dur=160/160/1040 seq=27258 lw=748000
2012-09-29 19:22:57.397908 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=320/320/1040 seq=27259 lw=748160
2012-09-29 19:22:57.417913 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=480/480/1040 seq=27260 lw=748320
2012-09-29 19:22:57.437960 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=640/640/1040 seq=27261 lw=748480
2012-09-29 19:22:57.457912 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=800/800/1040 seq=27262 lw=748640
2012-09-29 19:22:57.477904 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=960/960/1040 seq=27263 lw=748800
2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2636 Send end packet for [2] ts=748000 dur=1120/1120/1040 seq=27264 lw=748800
2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2636 Send end packet for [2] ts=748000 dur=1120/1120/1040 seq=27265 lw=748800
2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2636 Send end packet for [2] ts=748000 dur=1120/1120/1040 seq=27266 lw=748800
2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2589 Queue digit delay of 40ms

Any idea would be greatly appreciated.

All the best,
Emrah
On Sep 27, 2012, at 3:54 PM, Emrah <lists at kavun.ch> wrote:

> Hey Ken,
> 
> I tried pass_rfc2833 with no noticeable change in the delay. It seemed to have made it less accurate though, especially in fast speed sequences.
> 
> Can I debug this further and how?
> 
> Thanks!
> On Sep 27, 2012, at 3:00 PM, Ken Rice <krice at freeswitch.org> wrote:
> 
>> There can be a delay of DTMF in and DTMF out if you are sending long DTMFs
>> using 2833, FreeSWITCH gets the entire DMTF and duration then regenerates
>> it...
>> 
>> If you don't need to interpret the DTMF you can set a variable to make it
>> just pass the DTMF through untouched... But this has its own set of caveats
>> (ie: if whatever is sending you DTMF is broken it just pass broken 2833
>> DTMF)
>> 
>> See http://wiki.freeswitch.org/wiki/Variable_pass_rfc2833
>> 
>> 
>> 
>> On 9/27/12 1:49 PM, "Emrah" <lists at kavun.ch> wrote:
>> 
>>> MC, the issue does not happen with inband DTMF and there is no delay!
>>> 
>>> Any idea on how to debug this further? I can't use inband continuously.
>>> 
>>> Thanks!
>>> Emrah
>>> 
>>> On Sep 27, 2012, at 12:46 PM, Emrah <lists at kavun.ch> wrote:
>>> 
>>>> Never tried with inband DTMFs. Will check.
>>>> 
>>>> Thanks!
>>>> On Sep 27, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>> 
>>>>> Does it happen whether you use RFC2833 or inband DTMFs? Just curious.
>>>>> -MC
>>>>> 
>>>>> On Wed, Sep 26, 2012 at 3:44 PM, Emrah <lists at kavun.ch> wrote:
>>>>> Yes I did.
>>>>> BTW, the example in the Wiki contradicts the inline documentation in
>>>>> switch.xml.
>>>>>  <!--
>>>>>       The min-dtmf-duration specifies the minimum DTMF duration to use on
>>>>>       outgoing events. Events shorter than this will be increased in
>>>>> duration
>>>>>       to match min_dtmf_duration. You cannot configure a dtmf duration on
>>>>> a
>>>>>       profile that is less than this setting. You may increase this value,
>>>>>       but cannot set it lower than 400. This value cannot exceed
>>>>>       max-dtmf-duration. -->
>>>>> The Wiki shows an example with the value at 100.
>>>>> 
>>>>> I tried increasing and decreasing it to no avail, it does not seem to
>>>>> interfere with anything I can measure with my ear. :P
>>>>> On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com>
>>>>> wrote:
>>>>> 
>>>>>> You tried this:
>>>>>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
>>>>>> 
>>>>>> On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
>>>>>> Hi guys,
>>>>>> 
>>>>>> I am comparing this with an Asterisk and FreeSWITCH installation, using the
>>>>>> same route, same codecs, same carrier, same phones and same servers? :P
>>>>>> I experience a delay when pressing DTMFs on the line that uses FreeSWITCH.
>>>>>> I am estimating the delay to be around 500 ms.
>>>>>> 
>>>>>> What are the settings I can fine tune to avoid this?
>>>>>> 
>>>>>> All the 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
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>> 
>> -- 
>> 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
> 



From mike.burlingame at me.com  Sat Sep 29 22:39:53 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sat, 29 Sep 2012 11:39:53 -0700
Subject: [Freeswitch-users] TEST
In-Reply-To: <8445172F-65DE-4E2F-A346-DEF2BBD18E58@freeswitch.org>
References: <8445172F-65DE-4E2F-A346-DEF2BBD18E58@freeswitch.org>
Message-ID: <D67E36CE-2DB4-4112-9CF2-84AB6352A81B@me.com>

Looks good

Sent from my iPhone 4S

On Sep 29, 2012, at 11:28 AM, Brian West <brian at freeswitch.org> wrote:

> TEST
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sat Sep 29 23:31:06 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Sep 2012 12:31:06 -0700
Subject: [Freeswitch-users] TEST
In-Reply-To: <D67E36CE-2DB4-4112-9CF2-84AB6352A81B@me.com>
References: <8445172F-65DE-4E2F-A346-DEF2BBD18E58@freeswitch.org>
	<D67E36CE-2DB4-4112-9CF2-84AB6352A81B@me.com>
Message-ID: <CAMWiMy-an8YAPgYOdcWkMeB2DGeGjyXk6gmE6pGDdPca9gutJw@mail.gmail.com>

He's testing with his new toys :P

On Sat, Sep 29, 2012 at 11:39 AM, Mike Burlingame
<mike.burlingame at me.com> wrote:
> Looks good
>
> Sent from my iPhone 4S
>
> On Sep 29, 2012, at 11:28 AM, Brian West <brian at freeswitch.org> wrote:
>
>> TEST
>> --
>> Brian West
>> brian at freeswitch.org
>> FreeSWITCH Solutions, LLC
>> PO BOX PO BOX 2531
>> Brookfield, WI 53008-2531
>> Twitter: @FreeSWITCH_Wire
>> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
>> iNUM: +883 5100 1420 9266
>> UK: +44 20 3298 4900
>> ISN: 410*543
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From b2m at a-cti.com  Sun Sep 30 05:46:27 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Sun, 30 Sep 2012 07:16:27 +0530
Subject: [Freeswitch-users] CDR for fail2ban
In-Reply-To: <CALNZuwn-gmPoFJuVON=SZqiSZxjUy5+4S3M693COKqBYF+kYDw@mail.gmail.com>
References: <CAP2fbbeMu0DUAG5zfnmup71ye50K9UHELDV57O=M3JYraZFtQA@mail.gmail.com>
	<CALNZuwn-gmPoFJuVON=SZqiSZxjUy5+4S3M693COKqBYF+kYDw@mail.gmail.com>
Message-ID: <CAP2fbbfkooHSNe2WM=SxFK4T+4gEG6kP+JJYRt0td8oLXQQdXw@mail.gmail.com>

Thanks for the link, Also I wanted to look for outbound numbers and
duration to perform some action.

Like blocking trail users if they try to use international minutes like
more 30 mins or so.

Thanks,
Bala

On Sat, Sep 29, 2012 at 10:05 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Yes, there is an unmerged mod_fail2ban you can use, which I just doc'ed
> for you.
> Please update the wiki with more information as you test it.
>
> Once this has been tested in production, it would be great to merge it
> into stable...
>
> http://wiki.freeswitch.org/wiki/Mod_fail2ban
>
> -Avi Marcus
> BestFone
>
> On Sat, Sep 29, 2012 at 1:28 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>
>> Is it possible to create separate cdr log file to setup fail2ban.
>>
>> Thanks,
>> Bala
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Sep 30 08:05:28 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 30 Sep 2012 00:05:28 -0400
Subject: [Freeswitch-users] DTMF delay when using FreeSWITCH
In-Reply-To: <06F123E2-F649-4859-9675-26DBEE6FE68F@kavun.ch>
References: <CC8A0C72.54439%krice@freeswitch.org>
	<EDBB9235-D08F-4FEC-A577-1D516FE6AA8F@kavun.ch>
	<06F123E2-F649-4859-9675-26DBEE6FE68F@kavun.ch>
Message-ID: <2ED66FE3-BD07-4F5A-9B4A-907581444574@kavun.ch>

When I try this, the delay disappears but FS detects multiple DTMFs where I only send one?
<action application="export" data="dtmf_type=none" inline="true"/>
<action application="start_dtmf" />

start_dtmf_generate adds the delay and so I'm stuck? 

Any suggestion?
On Sep 29, 2012, at 2:29 PM, Emrah <lists at kavun.ch> wrote:

> Hi all,
> 
> I am now trying to force inband DTMF on my PSTN peers alone. I tried the following with no luck:
> <action application="start_dtmf" />
> <action application="export" data="dtmf_type=inband" />
> 
> The delay is still there and I get the following output in my console for a single DTMF:
> 
> 2012-09-29 19:22:57.357913 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 2:1040
> 2012-09-29 19:22:57.357913 [DEBUG] switch_ivr_bridge.c:393 Send signal sofia/external/1234567890 [BREAK]
> 2012-09-29 19:22:57.377912 [DEBUG] switch_rtp.c:2736 Send start packet for [2] ts=748000 dur=160/160/1040 seq=27258 lw=748000
> 2012-09-29 19:22:57.397908 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=320/320/1040 seq=27259 lw=748160
> 2012-09-29 19:22:57.417913 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=480/480/1040 seq=27260 lw=748320
> 2012-09-29 19:22:57.437960 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=640/640/1040 seq=27261 lw=748480
> 2012-09-29 19:22:57.457912 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=800/800/1040 seq=27262 lw=748640
> 2012-09-29 19:22:57.477904 [DEBUG] switch_rtp.c:2636 Send middle packet for [2] ts=748000 dur=960/960/1040 seq=27263 lw=748800
> 2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2636 Send end packet for [2] ts=748000 dur=1120/1120/1040 seq=27264 lw=748800
> 2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2636 Send end packet for [2] ts=748000 dur=1120/1120/1040 seq=27265 lw=748800
> 2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2636 Send end packet for [2] ts=748000 dur=1120/1120/1040 seq=27266 lw=748800
> 2012-09-29 19:22:57.497915 [DEBUG] switch_rtp.c:2589 Queue digit delay of 40ms
> 
> Any idea would be greatly appreciated.
> 
> All the best,
> Emrah
> On Sep 27, 2012, at 3:54 PM, Emrah <lists at kavun.ch> wrote:
> 
>> Hey Ken,
>> 
>> I tried pass_rfc2833 with no noticeable change in the delay. It seemed to have made it less accurate though, especially in fast speed sequences.
>> 
>> Can I debug this further and how?
>> 
>> Thanks!
>> On Sep 27, 2012, at 3:00 PM, Ken Rice <krice at freeswitch.org> wrote:
>> 
>>> There can be a delay of DTMF in and DTMF out if you are sending long DTMFs
>>> using 2833, FreeSWITCH gets the entire DMTF and duration then regenerates
>>> it...
>>> 
>>> If you don't need to interpret the DTMF you can set a variable to make it
>>> just pass the DTMF through untouched... But this has its own set of caveats
>>> (ie: if whatever is sending you DTMF is broken it just pass broken 2833
>>> DTMF)
>>> 
>>> See http://wiki.freeswitch.org/wiki/Variable_pass_rfc2833
>>> 
>>> 
>>> 
>>> On 9/27/12 1:49 PM, "Emrah" <lists at kavun.ch> wrote:
>>> 
>>>> MC, the issue does not happen with inband DTMF and there is no delay!
>>>> 
>>>> Any idea on how to debug this further? I can't use inband continuously.
>>>> 
>>>> Thanks!
>>>> Emrah
>>>> 
>>>> On Sep 27, 2012, at 12:46 PM, Emrah <lists at kavun.ch> wrote:
>>>> 
>>>>> Never tried with inband DTMFs. Will check.
>>>>> 
>>>>> Thanks!
>>>>> On Sep 27, 2012, at 12:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>>> 
>>>>>> Does it happen whether you use RFC2833 or inband DTMFs? Just curious.
>>>>>> -MC
>>>>>> 
>>>>>> On Wed, Sep 26, 2012 at 3:44 PM, Emrah <lists at kavun.ch> wrote:
>>>>>> Yes I did.
>>>>>> BTW, the example in the Wiki contradicts the inline documentation in
>>>>>> switch.xml.
>>>>>> <!--
>>>>>>      The min-dtmf-duration specifies the minimum DTMF duration to use on
>>>>>>      outgoing events. Events shorter than this will be increased in
>>>>>> duration
>>>>>>      to match min_dtmf_duration. You cannot configure a dtmf duration on
>>>>>> a
>>>>>>      profile that is less than this setting. You may increase this value,
>>>>>>      but cannot set it lower than 400. This value cannot exceed
>>>>>>      max-dtmf-duration. -->
>>>>>> The Wiki shows an example with the value at 100.
>>>>>> 
>>>>>> I tried increasing and decreasing it to no avail, it does not seem to
>>>>>> interfere with anything I can measure with my ear. :P
>>>>>> On Sep 26, 2012, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com>
>>>>>> wrote:
>>>>>> 
>>>>>>> You tried this:
>>>>>>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF
>>>>>>> 
>>>>>>> On Wed, Sep 26, 2012 at 3:19 PM, Emrah <lists at kavun.ch> wrote:
>>>>>>> Hi guys,
>>>>>>> 
>>>>>>> I am comparing this with an Asterisk and FreeSWITCH installation, using the
>>>>>>> same route, same codecs, same carrier, same phones and same servers? :P
>>>>>>> I experience a delay when pressing DTMFs on the line that uses FreeSWITCH.
>>>>>>> I am estimating the delay to be around 500 ms.
>>>>>>> 
>>>>>>> What are the settings I can fine tune to avoid this?
>>>>>>> 
>>>>>>> All the 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
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>> 
>>> -- 
>>> 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
>> 
> 



From avi at avimarcus.net  Sun Sep 30 11:24:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 30 Sep 2012 09:24:53 +0200
Subject: [Freeswitch-users] CDR for fail2ban
In-Reply-To: <CAP2fbbfkooHSNe2WM=SxFK4T+4gEG6kP+JJYRt0td8oLXQQdXw@mail.gmail.com>
References: <CAP2fbbeMu0DUAG5zfnmup71ye50K9UHELDV57O=M3JYraZFtQA@mail.gmail.com>
	<CALNZuwn-gmPoFJuVON=SZqiSZxjUy5+4S3M693COKqBYF+kYDw@mail.gmail.com>
	<CAP2fbbfkooHSNe2WM=SxFK4T+4gEG6kP+JJYRt0td8oLXQQdXw@mail.gmail.com>
Message-ID: <CALNZuwkDi6+hKK+4FkePo-4AF5ey6ohR7Ba8oOTG351CC9wX7w@mail.gmail.com>

Ah. Well, with xml_cdr or json_cdr posting to a script, you can store the
relevant information in a way for fail2ban to use. However, I think that
would be a pretty hard pattern to come up to know there's fraud occurring
on an account.

There's a service/product from http://www.humbuglabs.org/ that analyzes
CDRs and uses some metric to determine if something is fraudulent, that
might be helpful for you. I haven't used it.

-Avi



On Sun, Sep 30, 2012 at 3:46 AM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Thanks for the link, Also I wanted to look for outbound numbers and
> duration to perform some action.
>
> Like blocking trail users if they try to use international minutes like
> more 30 mins or so.
>
> Thanks,
> Bala
>
>
> On Sat, Sep 29, 2012 at 10:05 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Yes, there is an unmerged mod_fail2ban you can use, which I just doc'ed
>> for you.
>> Please update the wiki with more information as you test it.
>>
>> Once this has been tested in production, it would be great to merge it
>> into stable...
>>
>> http://wiki.freeswitch.org/wiki/Mod_fail2ban
>>
>> -Avi Marcus
>> BestFone
>>
>> On Sat, Sep 29, 2012 at 1:28 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> Is it possible to create separate cdr log file to setup fail2ban.
>>>
>>> Thanks,
>>> Bala
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
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>>> 
>>> 
>>>
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>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing 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 maleki_farzaneh at yahoo.com  Sun Sep 30 09:16:54 2012
From: maleki_farzaneh at yahoo.com (Ms Farzaneh Maleki)
Date: Sat, 29 Sep 2012 22:16:54 -0700 (PDT)
Subject: [Freeswitch-users] FSClient
Message-ID: <1348982214.59368.YahooMailClassic@web110302.mail.gq1.yahoo.com>

I am a beginner in freeswitch. I installed an image from 2600hz.com that included CentOS, freeswitch and bluebox. 
I installed FSClient on 2 clients
I connect to freeswitch by using bluebox (ServerIp/bluebox) and set to device in it and assign numbers to them. 
After adding account in FSClient, when I dial the number of other client in fsclient, It does not work. I also active the SIP protocol in CentOS firewall
What should I do?
Regards,
Farzaneh
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From mitch.capper at gmail.com  Sun Sep 30 17:09:22 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sun, 30 Sep 2012 15:09:22 +0200
Subject: [Freeswitch-users] FSClient
In-Reply-To: <1348982214.59368.YahooMailClassic@web110302.mail.gq1.yahoo.com>
References: <1348982214.59368.YahooMailClassic@web110302.mail.gq1.yahoo.com>
Message-ID: <CALn233oJRZEV_Uhcz77Zc_OCFds8iCNLJ5em6YUsv6nE+zpXJA@mail.gmail.com>

First try dialing a server side number (like the tetris sound or similar)
or making an outbound call.   You will want to connect fs_cli to the server
and optionally the client (for FSClient it runs on port 8022 so fs_cli -P
8022) and set the mode to debug for the console (F8 I believe).  Then try
to make a call and see what happens.  If you can't figure out the issue
from the console log output use fs_logger.pl (search the wiki) i works just
like fs_cli but pastes the results on the pastebin and give you a link you
can reply back with so we can see the console output.

~Mitch


On Sun, Sep 30, 2012 at 7:16 AM, Ms Farzaneh Maleki <
maleki_farzaneh at yahoo.com> wrote:

> I am a beginner in freeswitch. I installed an image from 2600hz.com that
> included CentOS, freeswitch and bluebox.
> I installed FSClient on 2 clients
> I connect to freeswitch by using bluebox (ServerIp/bluebox) and set to
> device in it and assign numbers to them.
> After adding account in FSClient, when I dial the number of other client
> in fsclient, It does not work. I also active the SIP protocol in CentOS
> firewall
> What should I do?
> Regards,
> Farzaneh
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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