Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] directly mix 3 way voice


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
jengjr at gmail.com
Guest





PostPosted: Thu Aug 28, 2008 2:47 am    Post subject: [Freeswitch-users] directly mix 3 way voice Reply with quote

Hello :

Is it possible to directly mix 3 way voice ?
Not putting eny leg into holding music.

I found the script originate a session is quite different than an
agent call. Some channel attributes missing .

# calltest.js
n_sess = new Session() ;
res = n_sess.originate(n_sess, "sofia/inter2/1527%210.243.126.72" );

uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate
56089ae3-134c-4bfc-a697-b08194918bf2,2008-08-27
13:35:33,1219815333,sofia/inter2/1527,CS_SOFT_EXECUTE,FreeSWITCH,0000000000,,1527,,,,,PCMU,8000,PCMU,8000


ps:
Can someone kindly provide a "threeway" application sample .

Thanks !!

_______________________________________________
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
Back to top
brian at freeswitch.org
Guest





PostPosted: Thu Aug 28, 2008 6:50 pm    Post subject: [Freeswitch-users] directly mix 3 way voice Reply with quote

Didn't you say you're doing SIP? To not have it put into hold music
set the variable hold_music=silence and nobody will get music while
setting up the threeway call.

/b

On Aug 28, 2008, at 2:46 AM, Lee JJ wrote:

Quote:
Hello :

Is it possible to directly mix 3 way voice ?
Not putting eny leg into holding music.

Brian West
sip:brian@freeswitch.org




_______________________________________________
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
Back to top
jengjr at gmail.com
Guest





PostPosted: Tue Sep 02, 2008 4:18 am    Post subject: [Freeswitch-users] directly mix 3 way voice Reply with quote

Hello :

Yes , I am doing SIP.
I would like to do something like call jump , while 2 leg talking
NOT interrupt anyone leg , neither into music nor into silence.

Someone can originate another endpoint phone ring , and switch over .

It seems only use conference to achieve it.

It seems hard to transfer an already bridged call into conference room,
neither the originated call from scripts.


# calltest.js
n_sess = new Session() ;
res = n_sess.originate(n_sess, "sofia/inter2/1527%210.243.126.72" );

uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate
56089ae3-134c-4bfc-a697-b08194918bf2,2008-08-27
13:35:33,1219815333,sofia/inter2/1527,CS_SOFT_EXECUTE,FreeSWITCH,0000000000,,1527,,,,,PCMU,8000,PCMU,8000


Quote:
Didn't you say you're doing SIP? To not have it put into hold music
set the variable hold_music=silence and nobody will get music while
setting up the threeway call.

Quote:
/b

Quote:
Quote:
On Aug 28, 2008, at 2:46 AM, Lee JJ wrote:

Quote:
Quote:
Hello :

Is it possible to directly mix 3 way voice ?
Not putting eny leg into holding music.

Quote:
Quote:
Brian West

_______________________________________________
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
Back to top
anthony.minessale at g...
Guest





PostPosted: Tue Sep 02, 2008 10:40 am    Post subject: [Freeswitch-users] directly mix 3 way voice Reply with quote

you can use mod_conference to make the call to begin with if you want.
There is bridge emulation mode where conference app pretends to be the bridge app and sets up a dedicated conference.

Also you can make the call as usual and bind a transfer to a * key to move you into a conference.

for instance make *3 warp you and the guy you are talking to to a conference at ext 3000
<action application="bind_meta_app" data="3 a s transfer::-both 3000"/>



On Tue, Sep 2, 2008 at 4:16 AM, Lee JJ <jengjr@gmail.com (jengjr@gmail.com)> wrote:
Quote:
Hello :

Yes , I am doing SIP.
I would like to do something like call jump , while 2 leg talking
NOT interrupt anyone leg , neither into music nor into silence.

Someone can originate another endpoint phone ring , and switch over .

It seems only use conference to achieve it.

It seems hard to transfer an already bridged call into conference room,
neither the originated call from scripts.


# calltest.js
n_sess = new Session() ;
res = n_sess.originate(n_sess, "sofia/inter2/1527%210.243.126.72" );

uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate
56089ae3-134c-4bfc-a697-b08194918bf2,2008-08-27
13:35:33,1219815333,sofia/inter2/1527,CS_SOFT_EXECUTE,FreeSWITCH,0000000000,,1527,,,,,PCMU,8000,PCMU,8000


Quote:
Didn't you say you're doing SIP? To not have it put into hold music
set the variable hold_music=silence and nobody will get music while
setting up the threeway call.

Quote:
/b

Quote:
Quote:
On Aug 28, 2008, at 2:46 AM, Lee JJ wrote:

Quote:
Quote:
Hello :

Is it possible to directly mix 3 way voice ?
Not putting eny leg into holding music.

Quote:
Quote:
Brian West

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services