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：
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）。

fs_cli测试

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

   

     

       

       

     

   

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" ;tag=597445KmmcvXp

To:

Call-ID: ae7f5675-6520-1232-9784-525400a16654

CSeq: 60529533 INVITE

Contact:

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"

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。