c - menu file did not played when user play "*",Asterisk-11.5.1 Confbridge -

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dialout user pickuped/answer call , merge confbridge admin getting "ringtone" asterisk-11.5.1 confbridge . ?

expected : admin user (a 7002) ,of current conference dailout , invite user (b 7001) join confernece. b picked call , joined confbridge. , b should communincate each other , press "*" listen conf menu file.

originale: b can listen menu press "*"; can not talk b . press * ,but menufile did not played . getting "ringingtone". why, ?

c**onference bridge name           users  marked locked? ================================ ====== ====== ======== 1010101                               2      1 unlocked**  *cli> confbridge list 1010101  channel                       user profile     bridge profile   menu             callerid ============================= ================ ================ ================ ================ sip/7002-00000009                              default_bridge   conf-admin-sub-dialout7002              sip/7001-0000000a             default_user     default_bridge   conf-admin-sub-dialout7001     *cli> sip show channels  peer             user/anr         call id          format           hold     last message    expiry     peer       xxx.yyy.zzz.xxx   7001             1deffeb72b0f045  (ulaw)           no       tx: ack                    7001       xxx.yyy.zzz.xxx   7002             fd2d41c9-e39354  (ulaw)           no       tx: ack                    7002        ==========================================================================   *cli> sip show channel 65a218b00e4e389    * sip call   curr. trans. direction:  outgoing   call-id:                65a218b00e4e389f56c1327c684e8513@xyz.xyz.xyz.xyz:5060   owner channel id:       sip/7001-0000000c   our codec capability:   (ulaw|alaw)   non-codec capability (dtmf):   1   codec capability:   (ulaw)   joint codec capability:   (ulaw)   format:                 (ulaw)   t.38 support            no   video support           no   maxcallbr:              384 kbps   theoretical address:    xxx.yyy.zzz.xxx:5060   received address:       xxx.yyy.zzz.xxx:5060   sip transfer mode:      open   force rport:            yes   audio ip:               xyz.xyz.xyz.xyz (local)   our tag:                as420f4f04   tag:              864d22e793aa05b8i0   sip user agent:            username:               7001   peername:               7001   original uri:           sip:7001@xxx.yyy.zzz.xxx:5060   caller-id:              91xxxxxxxxxxxx   need destroy:           no   last message:           tx: ack   promiscuous redir:      no   route:                  <sip:7001@xxx.yyy.zzz.xxx:5060>   dtmf mode:              rfc2833   sip options:            (none)   session-timer:          inactive  =========================================================================== *cli> sip show channel fd2d41c9-e39354    * sip call   curr. trans. direction:  outgoing   call-id:                fd2d41c9-e3935429@xxx.yyy.zzz.xxx   owner channel id:       sip/7002-00000009   our codec capability:   (ulaw|alaw)   non-codec capability (dtmf):   1   codec capability:   (ulaw)   joint codec capability:   (ulaw)   format:                 (ulaw)   t.38 support            no   video support           no   maxcallbr:              384 kbps   theoretical address:    xxx.yyy.zzz.xxx:5061   received address:       xxx.yyy.zzz.xxx:5061   sip transfer mode:      open   force rport:            yes   audio ip:               xyz.xyz.xyz.xyz (local)   our tag:                as165d44ab   tag:              316d654987e586a9o1   sip user agent:         linksys/pap2t-3.1.15(ls)   username:               7002   peername:               7002   original uri:           sip:7002@xxx.yyy.zzz.xxx:5061   caller-id:              7002   need destroy:           no   last message:           tx: ack   promiscuous redir:      no   route:                  <sip:7002@xxx.yyy.zzz.xxx:5061>   dtmf mode:              rfc2833   sip options:               session-timer:          inactive ============================================================  *cli> sip show channelstats  peer             call id      duration recv: pack  lost       (     %) jitter send: pack  lost       (     %) jitter xxx.yyy.zzz.xxx   5e81a94e-44  00:03:51 0000010612  0000000000 ( 0.00%) 0.0000 0000009484  0000000000 ( 0.00%) 0.0006 xxx.yyy.zzz.xxx   65a218b00e4  00:02:17 0000006816  0000000000 ( 0.00%) 0.0000 0000006632  0000000000 ( 0.00%) 0.0006  ======================================================  cli> sip show channels peer             user/anr         call id          format           hold     last message    expiry     peer       xxx.yyy.zzz.xxx   7002             5e81a94e-449935  (ulaw)           no       tx: ack                    7002       xxx.yyy.zzz.xxx   7001             65a218b00e4e389  (ulaw)           no       tx: ack                    7001        ========================================================

i have solved ,by using ami , originate app .now works expected.


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