yes i would love to implement the announce conference count to user before entering but cant figure out how to "calculate" that. also. the announce count should work because i have tts installed. so i am still concerned why the whole room announce does not work. thanks.

From: "freeswitch-users-request@lists.freeswitch.org" <freeswitch-users-request@lists.freeswitch.org>
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, January 17, 2012 8:27 PM
Subject: FreeSWITCH-users Digest, Vol 67, Issue 159

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Today's Topics:

  1. FreeSWITCH Community Conference Call Tomorrow (Michael Collins)
  2. announce count mod conference (Rodney)
  3. Re: announce count mod conference (Michael Collins)
  4. Freeswitch & double NAT Configuration (Was:
      USER_NOT_REGISTERED with external profile)
      (Alexis de BRUYN [Mailinglists])
  5. Re: Play media dialplan (dan@subformat.net)
  6. Re: sip_auto_answer auto copy to bleg (Seven Du)
Hello all!

Just a reminder that we have our FreeSWITCH community conference call tomorrow. We have Moc scheduled to discuss some of the new things he's been cooking up. However, he might have to go play with his cable modem for a bit so we'll see how the schedule pans out. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_18

Make a note of the upcoming conference calls. Normally I could say that we are booked solid through February, but since 2012 is a leap year we actually have a call scheduled for February 29. I'd love to hear some suggestions for a special leap-year edition of the FS community conference call! Let me know what you think. :)

Thanks,
Michael
the following section of mod conference does not seem to work  for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that. 

I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers. 


for
announce-countRequires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached.5
I'm pretty sure that this is for the whole room. You are probably better off calculating the number of users in the conference just prior to sending them in:

http://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline

HTH,
MC

On Tue, Jan 17, 2012 at 11:32 AM, Rodney <notlikeme75@yahoo.com> wrote:
the following section of mod conference does not seem to work  for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that. 

I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers. 


for
announce-countRequires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached. 5

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Hi Everybody,

I am trying to get a conversation between two natted clients and a
natted freeswitch server. I am playing with a fresh install, the
external profile and default directory (1000 & 1001 users).

I can see that my clients are registered on the server logs (on the 5080
port). All others needed ports are opened too.

I have followed the instructions from the wiki to get a double nat
configuration, but my client phone are not ringing (and I cannot hear
anything).

All is working fine with defaults in a non nat configuration (with
internal profile).

Is there anything special to set up in the directory / user configuration ?

Can anyone give me some hints with his natted configuration ? I am stuck.

Thanks for your help.

Regards,

-------- Original Message --------
Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile
Date: Mon, 16 Jan 2012 23:37:20 +0100
From: Alexis de BRUYN [Mailinglists] <alexis.mailinglist@de-bruyn.fr>
Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
To: freeswitch-users@lists.freeswitch.org

Hi Everybody,

I am trying to use FreeSwitch in a (double) NAT Configuration from a
fresh snapshot install (Debian Squeeze, server outside from my client
LAN) with default directory for users.

When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the
callee, the phone doesn't ring and automatically hangup, I see in the
console :

2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]

However, 1000 and 1001 are registered (from the same LAN) :

sofia status profile external

=================================================================================================
Name                external
Domain Name          N/A
Auto-NAT            false
DBName              sofia_reg_external
Pres Hosts         
Dialplan            XML
Context              public
Challenge Realm      auto_to
RTP-IP              192.168.1.6
SIP-IP              192.168.1.6
URL                  sip:mod_sofia@192.168.1.6:5080
BIND-URL            sip:mod_sofia@192.168.1.6:5080
HOLD-MUSIC          local_stream://moh
OUTBOUND-PROXY      N/A
CODECS IN            G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM
CODECS OUT          PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO          0
MAX-DIALOG          0
NOMEDIA              false
LATE-NEG            false
PROXY-MEDIA          false
AGGRESSIVENAT        false
STUN-ENABLED        true
STUN-AUTO-DISABLE    false
CALLS-IN            3
FAILED-CALLS-IN      0
CALLS-OUT            0
FAILED-CALLS-OUT     0

Registrations:
=================================================================================================
Call-ID:        aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm
User:          1001@X.Y.Z.T
Contact:        "Freeswitch" <sip:1001@A.B.C.D:50193>
Agent:          Bria iOS 2.0.0
Status:        Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934)
Host:          phone
IP:            A.B.C.D
Port:          50193
Auth-User:      1001
Auth-Realm:     X.Y.Z.T
MWI-Account:    1001@X.Y.Z.T

Call-ID:        3c27b7424837-bsfi877ujb2n
User:          1000@X.Y.Z.T
Contact:        "freeswitch" <sip:1000@192.168.0.69:2048;line=skgk4hug>
Agent:          snom300/8.4.32
Status:        Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658)
Host:          phone
IP:            A.B.C.D
Port:          62061
Auth-User:      1000
Auth-Realm:     X.Y.Z.T
MWI-Account:    1000@X.Y.Z.T

Total items returned: 2
=================================================================================================

All necessary ports are opened/forwarded on the server.

I See on the 1000 configuration that this is the local ip address which
is set as contact. Is there any other setups to do in the directory ?
Or other parameters in the external profile ?

Thanks for your help !

Regards,

--
Alexis de BRUYN
Mail : alexis.mailinglist@de-bruyn.fr

_________________________________________________________________________
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--
--
Alexis de BRUYN
Mail : alexis.mailinglist@de-bruyn.fr


Aaah sorry! That shows just how new I am to FS.

Heres the rather large log: http://pastebin.freeswitch.org/18151

Thanks again.

Dan.

> Oops, you forgot to enable debug. It looks like you're capturing right
> from
> the console instead of using fs_cli. If you're at the console then type
> this before doing a test call:
>
> console loglevel debug
>
> Note: if you use fs_cli to connect to FS you'll automatically be at
> loglevel debug. Anyway, turn on debug level output and pastebin a new log.
>
> -MC
>
> On Tue, Jan 17, 2012 at 2:34 AM, <dan@subformat.net> wrote:
>
>> Thanks MC, Tried that but sleep doesn't appear to be having any effect?
>>
>> I have pasted my log here: http://pastebin.freeswitch.org/18143
>>
>> As you can see in the log the ext is not registered, but the dial plan
>> should continue on fail? to the media file etc.
>>
>> Thanks
>>
>> Dan.
>>
>>
>>
>> > On Mon, Jan 16, 2012 at 11:21 AM, Paul Cupis <paul@cupis.co.uk> wrote:
>> >
>> >> On 16/01/12 14:29, dan@subformat.net wrote:
>> >> > If you could point me in the right direction, that would be great.
>> >>
>> >> Can you provide a log of the call from FreeSWITCH, on
>> >> http://pastebin.freeswitch.org/ for us to look at, please?
>> >>
>> >>
>> > Also, you might want to add an action right after the answer:
>> > <action application="sleep" data="1500"/>
>> >
>> > Sometimes it takes a moment for media to "come up" and your playback
>> could
>> > occur before media is happily flowing. If you sleep for a short period
>> of
>> > time usually that helps. Try making the sleep duration longer or
>> shorter
>> > to
>> > see what happens.
>> >
>> > -MC
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting@freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> > http://www.cudatel.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users@lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting@freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




I use 7.60.0.110.rom btw.


On Wednesday, January 18, 2012 at 1:30 AM, Anthony Minessale wrote:
I heard on the beta firmware for yealink that they support the multicast paging as done in esf_page_group from mod_esf


On Tue, Jan 17, 2012 at 12:45 AM, Seven Du <dujinfang@gmail.com> wrote:
seems yealink phone has problem, see


there's no intercom in b-leg, but maybe because Call-Info present ?

   Call-Info: <sip:192.168.7.143>;answer-after=0





The following confirmed work

 originate {sip_auto_answer=true}user/1000 &bridge({sip_auto_answer=false,sip_h_Call-info='x'}user/1002)

   INVITE sip:1002@192.168.7.105:5062 SIP/2.0
   Call-Info: <x>


On Tuesday, January 17, 2012 at 8:44 AM, Anthony Minessale wrote:

originate [sip_auto_answer=true]user/1000 &bridge([sip_auto_answer=false]user/1001)


On Mon, Jan 16, 2012 at 1:15 PM, Seven Du <dujinfang@gmail.com> wrote:
Hi,

I use   originate {sip_auto_answer}user/1000 &bridge(user/1001) and want 1000 to be auto answered but not 1001, but it seems FS automatically copy to b-leg.  I manually comment the following lines and it seems to work.



// switch_channel_set_variable(nchannel, "sip_invite_params", "intercom=true");
}

switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER);
// switch_ivr_transfer_variable(session, nsession, "sip_auto_answer");

would it be good to default to not copy but leave the "export" or equivalent to do that? Or should I patch to add a var to disable that?

Thanks.

-- 

Sent with Sparrow


_________________________________________________________________________
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--
Anthony Minessale II

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Twitter: http://twitter.com/FreeSWITCH_wire

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_________________________________________________________________________
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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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@hotmail.com
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
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_________________________________________________________________________
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