Question

我正在尝试为给定号码创建传入/传出的拨号方案：

+ xx xxx [xxxxxxxxx | xxxxxxxx]

我已在sip.conf

中配置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，我想创建传入/传出中继，我的分机允许拨打国际电话和接收到的给定号码的来电。

+ xx xxx [xxxxxxxxx | xxxxxxxx]

[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：通话.... 然后，VM播放：Person you are calling is unavailable

Asterisk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 78a9b28011fd522601047c9317adca91@xx.xx.xx.xx: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 78a9b28011fd522601047c9317adca91@xx.xx.xx.xx: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...

Answer 1

基本上拨号串可以在SIP / devicename / extension＆＃39;或者＆＃39; SIP / username @ host＆＃39;格式。 SIP/sipprovider/0091XXXXX99999@sipprovider错了。

＆＃34;达到重传超时＆＃34;意味着星号尝试将一个INVITE发送给sipprovider，但是sipprovider的SIP端口（5060 UDP）是不可访问的。您可以在SIP调试中看到这一点。

如何更新我的Asterisk Dial-Plan以结合电话号码？

1 个答案: