我有一个使用freeswitch设置的sip配置文件
<include>
<gateway name="[sipname]">
<param name="register" value="false"/>
<param name="username" value="not-used"/>
<param name="password" value="not-used"/>
<param name="realm" value="[ip address]"/>
<param name="proxy" value="[ip address]"/>
</gateway>
</include>
我是否正确地认为我应该可以拨打电话:
originate sofia/external/[phonenumber]@[ip address] '&javascript(/root/voice.js 20)'
因为我一直收到RECOVERY_ON_TIMER_EXPIRE,在我调查所有其他问题之前,我可能想检查一下到目前为止我做得对。
答案 0 :(得分:2)
拨出网关
基本形式:
*sofia/gateway/<gateway>/<number_to_dial>*
示例1:
*sofia/gateway/asterlink/18005551212*
http://wiki.freeswitch.org/wiki/Mod_sofia#Dial_out_of_a_gateway
originate sofia/gateway/[sipname]/[phonenumber] &javascript('/root/voice.js 20')
答案 1 :(得分:0)
originate {ignore_early_media=true,bridge_early_media=false,originate_timeout=24,call_timeout=24,execute_on_answer='sched_hangup 10 alloted_timeout',origination_uuid=f7f927c5-n6wv-a55j-m5cr-37c9-f7f934d95e,origination_caller_id_number=+12121112222}sofia/gateway/asterisk/+14171111111 handle_calls
在这种情况下,星号是/sip_profiles/external/asterisk.xml
> sofia status gateway asterisk
=================================================================================================
Name asterisk
Profile external
Scheme Digest
Realm 192.241.203.11:5090
Username FreeSWITCH
Password no
From <sip:FreeSWITCH@192.241.203.21:5090>
Contact <sip:gw+asterisk@2.67.78.1:5090;transport=udp;gw=asterisk>
Exten FreeSWITCH
To sip:FreeSWITCH@192.241.203.11:5090
Proxy sip:192.241.203.11:5090
Context public
Expires 3600
Freq 3600
Ping 0
PingFreq 0
PingTime 0.00
PingState 0/0/0
State NOREG
Status UP
Uptime 290s
CallsIN 0
CallsOUT 2
FailedCallsIN 0
FailedCallsOUT 0
答案 2 :(得分:0)
谈论FreeSWITCH而不是Asterisk。拨号命令不正确 - 通过网关应该是:
fs_cli&GT;发起sofia / external / [phonenumber] @ [网关名称]'&amp; yourscript'
答案 3 :(得分:-1)
首先运行fs_cli和命令&#34; sofia status&#34;检查网关是否已启动