Freeswitch与外域IP对接(outbound sip server)之outgoing call
专注三网融合之接入网/核心网等技术，音视频信令和媒体传输相关协议 专家前辈入驻， 谈笑有鸿儒，往来无白丁，共同学习探讨IMS/IPPBX相关技术 欢迎同行加入IMS/IPPBX/VoIP/SIP QQ交流群:255404741(猛戳直接加群) Precondition说明：fs_cli测试
本域：imsclub.com/122.112.86.102: 5060
测试分机：sip:[email protected]
外域：42.96.203.28:5060
测试分机：sip:[email protected]
目的：实现注册在imsclub.com/122.112.86.102的分机可以直接呼叫注册在42.96.203.28上的分机，例如1010拨进行直接呼叫88885000（42.96.203.28的服务器需要配置相应的sip route到imsclub.com）。
originate sofia/122.112.86.102/[email protected] 1010
呼叫 外域的sip:[email protected] 然后bridge sip:[email protected] 到本域的extension [email protected]
受此启发，可以写个dialplan，支持outbound sip server，并且通过oubound sip server直接呼出到其他域的分机，从而不需要以SIP Trunk的方式实现outgoing call。（当然对应的outbound sip server 要配置成接受外域对接的incoming call）
dialplan配置
conf/dialplan/default/下增加
00_ipconn_outbound_to_zhangwenge.xml
<!--- Added by vinco zhang at 2014-06-02 for ip connect outbound sip server -->
<include>
<extension name="extension-zhangwenge">
<condition field="destination_number" expression="^8888(\d+)$">
<action application="set" data="dialed_extension=8888$1"/>
<action application="bridge" data="sofia/internal/[email protected]"/>
</condition>
</extension>
</include>
dialplan配置生效
freeswitch@internal> reloadxml
呼叫测试
1. 确保本次测试不受SIP Trunk的干扰
除了无效的example.com 没有其他gateway：
freeswitch@internal> sofia status
Name Type Data State
=======================================================================================
external profile sip:[email protected]:5080 RUNNING (0)
external::example.com gateway sip:[email protected] NOREG
122.112.86.102 alias internal ALIASED
internal profile sip:[email protected]:5060 RUNNING (0)
internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)
=======================================================================================
3 profiles 1 alias
2. 1010拨88885000直接呼叫（注意不是输入sip uri [email protected]）
88885000话机接起，这样本域对接的outgoing call 成功了！抓包确认包确实是从 122.112.86.102:5060-->42.96.203.28:5060(而不是通过sip trunk 122.112.86.102:5080到达42.96.203.28:5060)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 122.112.86.102;rport;branch=z9hG4bKvFjNt4209Q00m
Max-Forwards: 69
From: "Extension 1010" <sip:[email protected]>;tag=597445KmmcvXp
To: <sip:[email protected]>
Call-ID: ae7f5675-6520-1232-9784-525400a16654
CSeq: 60529533 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: FreeSWITCH-mod_sofia/1.2.23~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 207
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
X-FS-Support: update_display,send_info
P-Asserted-Identity: "Extension 1010" <sip:[email protected]>
v=0
o=FreeSWITCH 1401709741 1401709742 IN IP4 122.112.86.102
s=FreeSWITCH
c=IN IP4 122.112.86.102
t=0 0
m=audio 21454 RTP/AVP 8 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
鸣谢：感谢 张文阁 前辈提供对接的sip服务器42.96.203.28
转载请注明出处 http://blog.csdn.net/xuyunzhang/article/details/28120903 谢谢。