retStr += "From: {uid} <sip:{uid}@{remote_ip}:{remote_port}>;tag={call_number}\r\n" retStr += "To: {uid} <sip:{uid}@{remote_ip}:{remote_port}>\r\n" retStr += "Call-ID: {call_id}\r\n" retStr += "User-Agent: fs testing\r\n" retStr += "CSeq: %d REGISTER\...
REGISTER sip:192.168.168.85SIP/2.0 Via:SIP/2.0/UDP192.168.168.168:;branch=z9hG4bK-d87543-6371fe0e115be271---d87543-;rport Max-Forwards: Contact:<sip:@192.168.168.168:;rinstance=196b0ce810f2e6f5> To:""<sip:@192.168.168.85> From:""<sip:@192.168.168.85>;tag=2d1fbf20 Call-ID:ZTRiYT...
Contact: <sip:alice@192.168.4.4:5090;rinstance=d42207a765c0626b;transport=UDP> To: <sip:alice@192.168.4.4;transport=UDP> From: <sip:alice@192.168.4.4;transport=UDP>;tag=9c709222 Call-ID: NmFjNzA3MWY1MDI3NGViMjY1N2QwZDlmZWQ5ZGY2OGE. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK,...
REGISTER sip:[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] <sip:[field0]@[remote_ip]:[remote_port]>;tag=[call_number] To: [field0] <sip:[field0]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 REGISTER Contac...
b:在向rtpproxy发送命令时,将extra_id_pv变量的值附加到Call-ID,必须设置extra_id_pv变量,注意:b不能与1、2、3同时使用 l:强制“查找”,只有当相应的会话在rtpproxy中已经存在时才重写SDP i/e:标志指定SIP消息的方向。i”表示内部网络(LAN),“e”表示外部网络(WAN)。使用时必须指定两个标志来定义传入网络和...
Call-ID: [call_id] Cseq: [cseq] BYE Contact: sip:[local_ip]:[local_port] Max-Forwards: 70 Content-Length: 0]]></send> 其中,$3 $4 $5 是变量,通过以下语法获取: <eregregexp="sip:(.*)>"search_in="hdr"header="Contact"assign_to="4,5"/><assignassign_to="4"variable="5"/> ...
INVITE sip:1003@10.11.15.109 SIP/2.0 Via: SIP/2.0/UDP10.12.0.1:21812;branch=z9hG4bK-d8754z-895ef25f292f2227-1---d8754z-;rport Max-Forwards: 70 Contact: To:"1003" From:"1000";tag=6328494c Call-ID: YWRiYzdjN2JlYWU0M2U0OWZkZDVjNTYwMGU0ZTYxMDg. ...
Call-Id: 1-reg@191.169.150.251 Cseq: 2762 REGISTER Contact: sip:6540012@191.169.150.251 Expires: 100 Content-Length: 0 Accept-Language: en Supported: sip-cc, sip-cc-01, timer User-Agent: Pingtel/1.2.7 (VxWorks) Via: SIP/2.0/UDP 191.169.150.251 ...
sip_call_id sip_call_idCreated by Ryan Harris, last modified on 2018.02.07sip_call_idstring SIP header Call-ID. Edit this page Previous sip_auto_simplify Next sip_callee_id_name sip_call_id Video Youtube Community Forums Twitter More Cluecon GitHub...
它支持SIP、H323、Skype、Google Talk等协议,并能很容易地与各种开源的PBX系统如sipXecs、Call Weaver、Bayonne、YATE及Asterisk等通信。FreeSWITCH 遵循RFC并支持很多高级的SIP特性,如 presence、BLF、SLA以及TCP、TLS和sRTP等。它也可以用作一个SBC进行透明的SIP代理(proxy)以支持其它媒体如T.38等。FreeSWITCH 支持...