背景信息：
我正在尝试按照此处的教程进行操作： http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc
基本设置如下：
- 两部 Polycom 电话（192.168.1.100 和 192.168.1.102）
- 处理注册和用户位置的 Kamailio 服务器 (192.168.1.101)
- 负责视频邮件、电话会议等的 Freeswitch 服务器 (192.168.1.111)
问题：
坏消息是，当我尝试从 999 拨打分机 888 或反之时，它不起作用。我听到忙音，尽管两部电话都在线。（值得一提的是，电话会议有效）
代码/配置：
这是默认容器/开箱即用的拨号计划中的部分，已添加以支持我的配置”
<extension name="kbridge">
<condition field="destination_number" expression="^(.+)$">
<action application="set" data="proxy_media=true"/>
<action application="set" data="call_timeout=50"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="sip_invite_domain=192.168.1.111"/>
<action application="export" data="sip_contact_user=ufs"/>
<action application="bridge" data="sofia/$${domain}/[email protected]"/>
<action application="answer"/>
<action application="voicemail" data="default ${domain_name} $1"/>
</condition>
</extension>
这是我的 acl.conf.xml 的部分内容：（请注意，我添加了我的 Kam 服务器的 IP 地址……不确定这是否是我应该做的，或者添加我的 freeswitch 服务器的地址）
<list name="domains" default="deny">
<!-- domain= is special it scans the domain from the directory to build the ACL -->
<node type="allow" domain="$${domain}"/>
<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
<node type="allow" cidr="192.168.1.101"/>
<node type="allow" cidr="192.168.1.111"/>
</list>
这是公共图表（public.xml）
<extension name="from_kamailio">
<condition field="network_addr" expression="^192\.168\.1\.101$" />
<condition field="destination_number" expression="^(.+)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
我目前所做的/调试信息
在我的 kamailio 服务器（端口 5060）上使用 tcpdump，我可以看到呼叫确实到达了 freeswitch。但 freeswitch 不会将呼叫发送回代理，然后再发送回电话。它返回以下 sip 消息：
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK15fa.79754276.0
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKb403de76A8D7964
Max-Forwards: 14
From: "999" <sip:[email protected]>;tag=C73F8AE1-87260556
To: <sip:[email protected];user=phone>;tag=KB5jNeBv3ZmaQ
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140320T233219Z~dd242f3ba6~32bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Remote-Party-ID: "888" <888>;party=calling;privacy=off;screen=no
当我查看 freeswitch 客户端中的调试数据时，我看到了以下内容：此呼叫：（仅发布了关键项目）
2014-03-26 17:02:31.825479 [NOTICE] switch_channel.c:1053 New Channel sofia/external/[email protected] [f284cfd6-b529-11e3-aa7f-35418c26e85f]
2014-03-26 17:02:31.825479 [DEBUG] switch_channel.c:2177 (sofia/external/[email protected]) Callstate Change DOWN -> RINGING
2014-03-26 17:02:31.825479 [DEBUG] switch_core_state_machine.c:523 (sofia/external/[email protected]) State ROUTING
2014-03-26 17:02:31.825479 [DEBUG] mod_sofia.c:123 sofia/external/[email protected] SOFIA ROUTING
2014-03-26 17:02:31.825479 [DEBUG] switch_core_state_machine.c:164 sofia/external/[email protected] Standard ROUTING
2014-03-26 17:02:31.825479 [INFO] mod_dialplan_xml.c:558 Processing 999 <999>->888 in context public
Dialplan: sofia/external/[email protected] parsing [public->unloop] continue=false
Dialplan: sofia/external/[email protected] Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/[email protected] Regex (PASS) [from_kamailio] network_addr(192.168.1.101) =~ /^192\.168\.1\.101$/ break=on-false
Dialplan: sofia/external/[email protected] Regex (PASS) [from_kamailio] destination_number(kb-888) =~ /^(.+)$/ break=on-false
Dialplan: sofia/external/[email protected] Action transfer(888 XML default)
2014-03-26 17:02:31.825479 [NOTICE] switch_ivr.c:1837 Transfer sofia/external/[email protected] to XML[888@default]
2014-03-26 17:02:31.825479 [INFO] mod_dialplan_xml.c:558 Processing 999 <999>->888 in context default
2014-03-26 17:02:31.845470 [NOTICE] switch_ivr.c:1837 Transfer sofia/external/[email protected] to enum[888@default]
2014-03-26 17:02:31.845470 [DEBUG] switch_core_state_machine.c:530 (sofia/external/[email protected]) State EXECUTE going to sleep
2014-03-26 17:02:31.845470 [DEBUG] switch_core_state_machine.c:467 (sofia/external/[email protected]) Running State Change CS_ROUTING
2014-03-26 17:02:31.845470 [DEBUG] switch_core_state_machine.c:523 (sofia/external/[email protected]) State ROUTING
2014-03-26 17:02:31.845470 [DEBUG] mod_sofia.c:123 sofia/external/[email protected] SOFIA ROUTING
2014-03-26 17:02:31.845470 [DEBUG] switch_core_state_machine.c:164 sofia/external/[email protected] Standard ROUTING
2014-03-26 17:02:31.845470 [DEBUG] mod_enum.c:642 ENUM Lookup on kb-888
2014-03-26 17:02:31.845470 [DEBUG] mod_enum.c:494 No Nameservers specified, using host default
2014-03-26 17:02:32.005525 [NOTICE] switch_core_state_machine.c:313 sofia/external/[email protected] has executed the last dialplan instruction, hanging up.
2014-03-26 17:02:32.005525 [NOTICE] switch_core_state_machine.c:315 Hangup sofia/external/[email protected] [CS_EXECUTE] [NORMAL_CLEARING]
然后它显示 SIP 暂时不可用错误消息。
我尝试按照教程操作，但不起作用。如有任何建议，我将不胜感激。谢谢。
编辑1
我不知道为什么这种解决方法有效......但我发现像这样改变拨号计划可以解决问题：
<extension name="kbridge">
<condition field="destination_number" expression="^(.+)$">
<action application="set" data="proxy_media=true"/>
<action application="set" data="call_timeout=50"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="sip_invite_domain=192.168.1.111"/>
<action application="export" data="sip_contact_user=ufs"/>
<!--<action application="bridge" data="sofia/$${domain}/[email protected]"/>-->
<action application="bridge" data="sofia/external/[email protected]"/>
<action application="answer"/>
<action application="voicemail" data="default ${domain_name} $1"/>
</condition>
</extension>
请注意，我已删除变量 $${domain} 并将其硬编码为“external”。现在我的扩展之间的调用可以正常工作。我不知道为什么会这样，但我目前正在寻找域变量的定义位置。