我正在尝试为给定号码的传入/传出创建拨号计划:
+xx xxx[xxxxxxx | xxxxxxx]
我已经在sip.conf中配置了我的服务提供商信息

[sipprovider]
type=friend
secret=xxxxx
defaultusername=xxxxx
host=xxx.xx.xx.xxx
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
insecure=port,invite
;fromdomain=xxx.xx.xx.xxx
context=default
nat=yes

现在,我要创建传入/传出中继,我的分机允许拨打国际长途和按给定号码接收的传入呼叫。
+xx xxx[xxxxxxx | xxxxxxx]
[default]
    switch => Realtime

    exten => 55,1,Verbose(1,Echo test application)
    exten => 55,n,Dial(SIP/sipprovider/0091XXXXX99999@sipprovider); Here is the outbound call, the exact dialstring depends on outgoing provider and channeltype
    exten => 55,n,Hangup()

显示:正在呼叫。。。。
然后,虚拟机播放:Person you are calling is unavailable
星号控制台:
== Using SIP RTP CoS mark 5
    -- Executing [55@default:1] Verbose("SIP/3001-00000029", "1,Echo test application") in new stack
 Echo test application
    -- Executing [55@default:2] Dial("SIP/3001-00000029", "SIP/sipprovider/0091XXXXX99999@sipprovider") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/sipprovider/0091XXXXX99999@sipprovider
[Aug 17 18:29:02] WARNING[32467]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Aug 17 18:29:02] WARNING[32467]: chan_sip.c:4053 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [55@default:3] Hangup("SIP/3001-00000029", "") in new stack
  == Spawn extension (default, 55, 3) exited non-zero on 'SIP/3001-00000029'
    -- Executing [h@default:1] Verbose("SIP/3001-00000029", "Hangup...") in new stack
Hangup...

最佳答案

基本上,dialstring可以是“SIP/devicename/extension”或“SIP/username@host”格式。SIP/sipprovider/0091XXXXX99999@sipprovider是错误的。
“已达到重新传输超时”表示星号试图向sipprovider发送邀请,但sipprovider的SIP端口(5060 UDP)不可访问。您可以在SIP调试中看到这一点。

关于linux - 如何为电话号码组合更新我的Asterisk拨号计划?,我们在Stack Overflow上找到一个类似的问题:https://stackoverflow.com/questions/32051406/

10-16 10:34