Any Help ? ........... Dialout user Pickuped/Answer call and merge into Confbridge but Admin getting "Ringtone" Asterisk-11.5.1 Confbridge . ?
Expected : admin user (A 7002) ,of current Conference Dailout and Invite user (B 7001) to join Confernece. B Picked call and joined Confbridge. A and B should Communincate with each other and press "*" to listen conf Menu file. Originale: B can listen menu by Press "*"; A can not Talk to B . A press * ,but MenuFile did not played . A only getting "Ringingtone". Why any help ? steps: 1: A 7002 adminuser ,start conference 1010101 2: A 7002 adminuser, Press "*" to listen menufile . 3: A 7002 adminuse , Press 5 to "Dialout" 4: A 7002 adminuse ,Entered "7001" to invited normal user to Join Conference 5: B 7001 user ,got ring 6: B 7001 user ,Pickup call and Join conference 1010101 7: B 7001 user ,Press * and Listen Menu file . "ok" 8: A 7002 admin, press * but , menufile did not played . "issue" 9: A 7002 admin, able to listen " ringing tone only " Conference 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: 65a218b00e4e389f56c1327c684e8...@xyz.xyz.xyz.xyz:5060 Owner channel ID: SIP/7001-0000000c Our Codec Capability: (ulaw|alaw) Non-Codec Capability (DTMF): 1 Their 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 Their Tag: 864d22e793aa05b8i0 SIP User agent: Username: 7001 Peername: 7001 Original uri: sip:7...@xxx.yyy.zzz.xxx:5060 Caller-ID: 91xxxxxxxxxxxx Need Destroy: No Last Message: Tx: ACK Promiscuous Redir: No Route: <sip:7...@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-e3935...@xxx.yyy.zzz.xxx Owner channel ID: SIP/7002-00000009 Our Codec Capability: (ulaw|alaw) Non-Codec Capability (DTMF): 1 Their 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 Their Tag: 316d654987e586a9o1 SIP User agent: Linksys/PAP2T-3.1.15(LS) Username: 7002 Peername: 7002 Original uri: sip:7...@xxx.yyy.zzz.xxx:5061 Caller-ID: 7002 Need Destroy: No Last Message: Tx: ACK Promiscuous Redir: No Route: <sip:7...@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 ======================================================== -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev