Question

有人可以指出我可以为可以容纳1或2部手机的测试设置获得正确配置的位置。

我在Ubuntu上使用N210和SBX子板在Asterisk 1.8.4上设置了一个OpenBTS 2.8。我能够拨打600并与BTS建立连接，并且echotest完美运行。我通过以下配置分配了连接到BTS的两个终端，当我尝试互相呼叫时，我收到下面发布的错误

调试输出显示它拨打了电话而我在另一部手机上没有响铃，我无法解除通话。它按预期超时。

这是我的extensions.conf

[macro-dialGSM]
exten => s,1,Dial(SIP/${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Busy(30)
exten => s-CONGESTION,1,Congestion(30)
exten => s-CHANUNAVAIL,1,playback(ss-noservice)
exten => s-CANCEL,1,Hangup
[sip-external]
exten => 9000,1,Macro(dialGSM,IMSI240020702009669)
exten => 9001,1,Macro(dialGSM,IMSI240016010357097)

这是我的sip.conf

[IMSI240020702009669]
callerid=9000
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info

[IMSI240016010357097]
callerid=9001
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info

这是星号

的错误输出

-- Executing [s@macro-dialGSM:1] Dial("SIP/IMSI240016010357097-0000001f","SIP/IMSI240020702009669,20") in new stack
== Using SIP RTP CoS mark 5
-- Called IMSI240020702009669
-- Nobody picked up in 20000 ms
-- Executing [s@macro-dialGSM:2] Goto("SIP/IMSI240016010357097-0000001f", "s-NOANSWER,1") in new stack
-- Goto (macro-dialGSM,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-dialGSM:1] Hangup("SIP/IMSI240016010357097-0000001f", "") in new stack
== Spawn extension (macro-dialGSM, s-NOANSWER, 1) exited non-zero on'SIP/IMSI240016010357097-0000001f' in macro 'dialGSM'
== Spawn extension (sip-external, 9000, 1) exited non-zero on'SIP/IMSI240016010357097-0000001f'
[Sep 18 18:01:31] WARNING[9737]: chan_sip.c:3551 retrans_pkt: Retransmission timeout reached on transmission 3c5b249c2220ff282dddf34d75e0848a@192.168.10.1:5060 for seqno 102(Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

你认为我在哪里弄错了？我提到了维基但它没有帮助，或者我无法理解如何从维基解决错误消息点。

Answer 1

我想出了宏必须通过ip来路由流量

的问题

Macro(dialGSM,IMSI240020702009669@127.0.0.1:5062)

希望这有助于某人

Answer 2

确实，为Dial功能提供ip地址/端口解决了我的问题。在我偶然发现这个解决方案之前，这是非常令人沮丧的。以下是正在运行的代码

sip.conf：

[IMSI3102XXXXXXXXXX3]
callerid=2000003
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info

[IMSI3102XXXXXXXXXX4]
callerid=2000004
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info

extentions.conf：

[macro-dialGSM]
exten => s,1,Dial(SIP/${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Busy(30)
exten => s-CONGESTION,1,Congestion(30)
exten => s-CHANUNAVAIL,1,playback(ss-noservice)
exten => s-CANCEL,1,Hangup
[sip-external]
exten => 2000003,1,Macro(dialGSM,IMSI3102XXXXXXXXXX3@127.0.0.1:5062)
exten => 2000004,1,Macro(dialGSM,IMSI3102XXXXXXXXXX4@127.0.0.1:5062)

使用OpenBTS调用未放置在Asterisk上

2 个答案: