From: r. <rao...@16...> - 2012-03-08 07:57:37
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How can i solve this problem ? beg for all answers . 2012-03-08 raojlist 发件人: raojlist 发送时间: 2012-03-08 15:52:19 收件人: sipp-users 抄送: 主题: [Sipp-users] How_can__i_hold_many_calls_from_other_agent,not noly one. Hi,all i want to continuity hold more than one calls from other agents, but use single following scenario xml file.. just can hold only one call. regards. ------------------------------------------ <?xml version="1.0" encoding="ISO-8859-1" ?> <scenario name="Basic UAS responder"> <recv request="INVITE" crlf="true"> </recv> <send> <![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Length: 0 ]]> </send> <send retrans="500"> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv request="ACK" optional="true" rtd="true" crlf="true"> <!--<action> <exec play_pcap_audio="g711a.pcap"/> </action>--> </recv> <recv request="BYE"> </recv> <send> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Length: 0 ]]> </send> <pause milliseconds="2000"></pause> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> </scenario> 2012-03-08 raojlist 发件人: raojlist 发送时间: 2012-03-07 10:24:13 收件人: sipp-users 抄送: 主题: [Sipp-users] Hello all, I want to batch register agents ,and then hold incoming call, but have problem. Hello all, I can register into asterisk successful,but can not hold incoming call from other agent client registered by eyebeam, my sipp xml config file describes. following: <?xml version="1.0" encoding="us-ascii"?> <scenario name="New_Call"> <send retrans="500"> <![CDATA[ REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: "[field0]" <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number] To: "[field0]" <sip:[field0]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 REGISTER Contact: sip:[field0]@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Expires:3600 Content-Length: 0 ]]> </send> <recv response="100" optional="true" /> <recv response="200" crlf="true" /> <send> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Length: 0 ]]> </send> <recv request="INVITE" crlf="true" /> <send> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000]]> </send> <recv request="ACK" crlf="true" /> <pause milliseconds="13000" /> <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <label id="1" /> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200" /> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000" /> </scenario> |