拨出用户选择/接听电话并合并到Confbridge中,但管理员获取"铃声" Asterisk-11.5.1 Confbridge。 ?
预计:管理员用户(A 7002),当前会议Dailout和邀请用户(B 7001)加入Confernece。 B接到电话并加入了Confbridge。 A和B应该相互通信并按下" *"收听conf菜单文件。
原文: B可以通过按" *"; A不能和B说话。 按*,但MenuFile没有播放。 只有获得" Ringingtone"。 为什么,任何帮助?
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
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: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
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: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
========================================================
答案 0 :(得分:0)
我已经解决了它,通过使用AMI和Originate app。现在按预期工作。