星号单向音频

Question

遇到一个奇怪的问题。尝试从 sip 客户端调用正常的 phone 或扩展。这总是导致单向音频连接。

我用的是odbc数据库，实在找不到问题所在。任何人都可以帮助我朝着正确的方向前进。似乎完全没有错误。

[general]
context=public
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0:15060
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
language=ja
externaddr=x.x.x.x
localnet=x.x.x.x/255.255.240.0
nat=force_rport,comedia
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm

/var/log/asterisk/messages

[Apr 12 10:44:36] VERBOSE[23055][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824@context_tok:1] NoOp("SIP/inbound_1_1-00000003", "inbound") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824@context_tok:2] Dial("SIP/inbound_1_1-00000003", "SIP/1_1_1_1/1_1_1_1&SIP/1_1_1_2/1_1_1_2&SIP/1_1_1_3/1_1_1_3&SIP/1_1_1_4/1_1_1_4") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_1/1_1_1_1
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_3/1_1_1_3
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_1-00000004 is ringing
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 is ringing
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 answered SIP/inbound_1_1-00000003
[Apr 12 10:44:44] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] pbx.c: Spawn extension (context_tok, 52431824, 2) exited non-zero on 'SIP/inbound_1_1-00000003'

试了好几种方法，在网上搜索，找不到正确的解决方案。

Answer 1

帮助以后遇到同样问题的人。

在 odbc 数据库中，我们有关于 null 的标准数据。

为数据库中的用户将 quality 设置为 yes 解决了这个问题。

星号单向音频

Asterisk one way audio

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