[Freeswitch-users] announce count mod conference
Rodney
notlikeme75 at yahoo.com
Wed Jan 18 04:54:32 MSK 2012
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.
________________________________
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To: freeswitch-users at 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 at 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-count Requires 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 at 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-count Requires 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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>
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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 at de-bruyn.fr>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
To: freeswitch-users at 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 at 192.168.1.6:5080
BIND-URL sip:mod_sofia at 192.168.1.6:5080
HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS IN G7221 at 32000h,G7221 at 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 at X.Y.Z.T
Contact: "Freeswitch" <sip:1001 at 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 at X.Y.Z.T
Call-ID: 3c27b7424837-bsfi877ujb2n
User: 1000 at X.Y.Z.T
Contact: "freeswitch" <sip:1000 at 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 at 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 at de-bruyn.fr
_________________________________________________________________________
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--
--
Alexis de BRUYN
Mail : alexis.mailinglist at 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 at 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 at cupis.co.uk> wrote:
>> >
>> >> On 16/01/12 14:29, dan at 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 at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> >
>> >
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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://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 at gmail.com> wrote:
>
>seems yealink phone has problem, see
>>
>>
>>http://pastebin.freeswitch.org/18138
>>
>>
>>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 at 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 at 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.
>>>>
>>>>--
>>>>About: http://about.me/dujinfang
>>>>Blog: http://www.dujinfang.com
>>>>Proj: http://www.freeswitch.org.cn
>>>>
>>>>Sent with Sparrow
>>>>
>>>>
>>>>
>>>>_________________________________________________________________________
>>>>Professional FreeSWITCH Consulting Services:
>>>>consulting at freeswitch.org
>>>>http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
>>>>Official FreeSWITCH Sites
>>>>http://www.freeswitch.org
>>>>http://wiki.freeswitch.org
>>>>http://www.cluecon.com
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>>--
>>>Anthony Minessale II
>>>
>>>FreeSWITCH http://www.freeswitch.org/
>>>ClueCon http://www.cluecon.com/
>>>Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>>AIM: anthm
>>>MSN:anthony_minessale at hotmail.com
>>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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
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