Yes, it is what I need. But now I have problem with sending dtmf. Here what I've done: [root@centos4-4-vm ~]# telnet localhost 8021 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted api originate user/1007@master.agent.starpoundtech.net &park() Content-Type: api/response Content-Length: 41 +OK bba3b45a-4cc1-48af-a15d-1052d5f11371 SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371 call-command: execute execute-app-name: eavesdrop execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c Content-Type: command/reply Reply-Text: +OK Here I had started to hear A and B. Here what I saw in FS log: 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/sip:1007@172.26.10.64:5060;fs_nat=yes [BREAK] 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/sip:1007@172.26.10.64:5060;fs_nat=yes Command Execute eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c) 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297 switch_core_media_bug_add() Attaching BUG to sofia/internal/1006@master.agent.starpoundtech.net 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1006@master.agent.starpoundtech.net receive message [TRANSCODING_NECESSARY] 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/sip:1007@172.26.10.64:5060;fs_nat=yes receive message [TRANSCODING_NECESSARY] Then I run command: api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1 Content-Type: api/response Content-Length: 14 -ERR no reply Log: 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190 switch_core_session_send_dtmf_string() sofia/internal/sip:1007@172.26.10.64:5060;fs_nat=yes send dtmf digit=1 ms=250 samples=2000 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet for [1] ts=2241760 dur=160/160/2000 seq=21346 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=320/320/2000 seq=21347 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=480/480/2000 seq=21348 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=640/640/2000 seq=21349 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=800/800/2000 seq=21350 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=960/960/2000 seq=21351 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1120/1120/2000 seq=21352 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1280/1280/2000 seq=21353 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1440/1440/2000 seq=21354 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1600/1600/2000 seq=21355 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1760/1760/2000 seq=21356 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1920/1920/2000 seq=21357 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21358 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21359 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21360 But both A and B couldn't hear me. Btw, after I had send dtmf "1" manually from my phone. B started to hear me. There was this record in log: 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2240 Does anybody know, what had I done wrong? ___ Thanks, Nikita
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Thursday, October 15, 2009 4:04 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] conference call
The wiki has a pretty good explanation of how eavesdrop works. Enabling a talk path to A or B or both A and B requires dtmf.
So, if C hits the 1 button on the phone they can talk to the UUID you bound the eavesdrop to. 2 talks to the other party. 3 talks to both parties.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
Adding more api calls to control eavesdrop via event socket should be possible, though I haven't looked at the code to see what it would take...
On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov@abisoft.spb.ru> wrote:
Thanks for answer.
As I've understood eavesdrop allows C to hear A and B, but A and B will not hear C. But in my case B SHOULD hear C. In my case A - client, B - agent worked with client and C - coach/supervisor for B. Can I make such conference in FS using eavesdrop or other API command?
___
Thanks, Nikita
________________________________
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Joao Mesquita Sent: Wednesday, October 14, 2009 7:38 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] conference call
Look at eavesdrop on the wiki.
JM
2009/10/14 Nikita Belov <nbelov@abisoft.spb.ru>
HI all,
I want to configure FS to make special conference call between three users (A, B, C). In this conference C will hear A and B, but A will hear only B. Can I make it using FS API commands? Does anybody know which approach is better to use?
___
Thanks, Nikita
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-- -Rupa
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