【发布时间】：2015-08-23 02:48:48
【问题描述】：
我正在尝试为给定号码的传入/传出创建拨号计划：
+xx xxx [xxxxxxxxx|xxxxxxxx]
我已经在sip.conf中配置了我的服务商信息
[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，我想创建呼入/呼出中继，我的分机允许拨打国际电话和给定号码收到的来电。
+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：呼叫....
然后，VM 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 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...
【问题讨论】：
标签： linux asterisk telecommunication